From daemonserj at gmail.com  Wed Aug  1 07:04:09 2012
From: daemonserj at gmail.com (daemonserj TVC)
Date: Wed, 1 Aug 2012 10:04:09 +0700
Subject: [Freeswitch-dev] mod_java is'nt loading (Debian x64)
Message-ID: <CAPd+SiC+JEFLMDDWd3zgNVPHdgvRzFJzdJO9SHXdVR5tD=FLbQ@mail.gmail.com>

Hello.
I've compiled & configured  freeswitch with option
--with-java=/path/to/jdk/include/ concerning
http://wiki.freeswitch.org/wiki/Mod_java
It successfuly compiled, but at start freeswitch writes to log
"2012-07-31 23:38:14.247729 [CRIT] switch_loadable_module.c:1310 Error
Loading module /usr/local/freeswitch/mod/mod_java.so
**Module load routine returned an error**

My java.conf.xml is :

<configuration name="java.conf" description="Java Plug-Ins">
  <!-- Path to the Java 1.6 virtual machine to use -->
  <javavm path="/usr/lib/jvm/java/jre/lib/i386/server/libjvm.so"/>
  <!-- Options to pass to Java -->
  <options>
    <!-- Your class path (make sure freeswitch.jar is on it) -->
    <option value="-Djava.class.path=$${base_dir}/scripts/freeswitch.jar"/> -->
    <!-- Enable remote debugging -->
    <option value="-agentlib:jdwp=transport=dt_socket,server=y,suspend=n,address=127.0.0.1:8000"/>
  </options>
  <startup class="net/cog/fs/system/Control" method="startup"
arg="start up arg"/>
  <shutdown class="net/cog/fs/system/Control" method="shutdown"
arg="shutdown arg"/>
</configuration>

libjvm.so is from stable debian package sun-java6-jdk (1.6.0_26). It's
successfuly linking.

I checked another JVM from oracle site (jdk-6u33-linux-x64.bin) and got this:
2012-07-31 23:38:14.247690 [ERR] modjava.c:211 Error loading
/usr/local/freeswitch/jdk1.6.0_33/jre/lib/amd64/server/libjvm.so

So I believe that problem is somewhere in mod_java but I have no more
debug messages to find out what is happened.
Please Help!


From peter.olsson at visionutveckling.se  Wed Aug  1 09:05:59 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 1 Aug 2012 05:05:59 +0000
Subject: [Freeswitch-dev] mod_java is'nt loading (Debian x64)
In-Reply-To: <CAPd+SiC+JEFLMDDWd3zgNVPHdgvRzFJzdJO9SHXdVR5tD=FLbQ@mail.gmail.com>
References: <CAPd+SiC+JEFLMDDWd3zgNVPHdgvRzFJzdJO9SHXdVR5tD=FLbQ@mail.gmail.com>
Message-ID: <tkeueitpcy40007iar4r0225.1343797389230@email.android.com>

http://jira.freeswitch.org.


/Peter

daemonserj TVC <daemonserj at gmail.com> skrev:
Hello.
I've compiled & configured  freeswitch with option
--with-java=/path/to/jdk/include/ concerning
http://wiki.freeswitch.org/wiki/Mod_java
It successfuly compiled, but at start freeswitch writes to log
"2012-07-31 23:38:14.247729 [CRIT] switch_loadable_module.c:1310 Error
Loading module /usr/local/freeswitch/mod/mod_java.so
**Module load routine returned an error**

My java.conf.xml is :

<configuration name="java.conf" description="Java Plug-Ins">
  <!-- Path to the Java 1.6 virtual machine to use -->
  <javavm path="/usr/lib/jvm/java/jre/lib/i386/server/libjvm.so"/>
  <!-- Options to pass to Java -->
  <options>
    <!-- Your class path (make sure freeswitch.jar is on it) -->
    <option value="-Djava.class.path=$${base_dir}/scripts/freeswitch.jar"/> -->
    <!-- Enable remote debugging -->
    <option value="-agentlib:jdwp=transport=dt_socket,server=y,suspend=n,address=127.0.0.1:8000"/>
  </options>
  <startup class="net/cog/fs/system/Control" method="startup"
arg="start up arg"/>
  <shutdown class="net/cog/fs/system/Control" method="shutdown"
arg="shutdown arg"/>
</configuration>

libjvm.so is from stable debian package sun-java6-jdk (1.6.0_26). It's
successfuly linking.

I checked another JVM from oracle site (jdk-6u33-linux-x64.bin) and got this:
2012-07-31 23:38:14.247690 [ERR] modjava.c:211 Error loading
/usr/local/freeswitch/jdk1.6.0_33/jre/lib/amd64/server/libjvm.so

So I believe that problem is somewhere in mod_java but I have no more
debug messages to find out what is happened.
Please Help!

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




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

Join Us At ClueCon - Aug 7-9, 2012

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

!DSPAM:50189a3b32762193214602!



From msc at freeswitch.org  Thu Aug  2 22:00:20 2012
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 2 Aug 2012 11:00:20 -0700
Subject: [Freeswitch-dev] The FreeSWITCH Devs - Let's Buy 'Em a Drink!
Message-ID: <CAKzWOxXrfUsacnNc0SexogND=y8-oqRhZb5rYwaRXbKEsRRn8A@mail.gmail.com>

Hey all!

It's ClueCon time and we'll all be in Chicago next week. As a token of
appreciation for all of their hard work we'd like to invite everyone to
donate a few dollars (or more!) in order to buy Anthony Minessale, Brian
West, and Mike Jerris a very well-deserved drink while they're in Chicago.
Simply click the Donate link on www.FreeSWITCH.org and add a note to seller
that this is for buying the developers a drink.

On a personal note I would just like to say that I have seen first hand
just how hard-working and dedicated these gentlemen really are. <hat tip/>
Thank you for making the FreeSWITCH project so awesome!

-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From amonroy at auronix.com  Tue Aug  7 00:25:25 2012
From: amonroy at auronix.com (Arturo monroy)
Date: Mon, 6 Aug 2012 15:25:25 -0500
Subject: [Freeswitch-dev] FreeSwitch Crashes When "Sending Bye before
	Blind_Transfer"
Message-ID: <007f01cd7411$9ccd2ad0$d6678070$@auronix.com>

Hey all!

 

                I've post a jira issue due FS crashes "appeared" after
sending bye before execute deflect command from Lua script, at jira issue
link I have attached logs (FS and windosws), a little analize and
screenshot. Does anyone have the same issue?

 

 <http://jira.freeswitch.org/browse/BKW-7>
http://jira.freeswitch.org/browse/BKW-7

 

regards arturo monroy

amonroy at auronix.com

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From anthony.minessale at gmail.com  Tue Aug  7 03:52:05 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 6 Aug 2012 18:52:05 -0500
Subject: [Freeswitch-dev] FreeSwitch Crashes When "Sending Bye before
	Blind_Transfer"
In-Reply-To: <007f01cd7411$9ccd2ad0$d6678070$@auronix.com>
References: <007f01cd7411$9ccd2ad0$d6678070$@auronix.com>
Message-ID: <CAKbxfG_9=j2F-xc952uJ=FR6F9LAvbaQ2VNfhYq9-e_czb0x=Q@mail.gmail.com>

why did you file it under BKW and not FreeSWITCH that is not a real
project so nobody will see your post.


On Mon, Aug 6, 2012 at 3:25 PM, Arturo monroy <amonroy at auronix.com> wrote:
> Hey all!
>
>
>
>                 I?ve post a jira issue due FS crashes ?appeared? after
> sending bye before execute deflect command from Lua script, at jira issue
> link I have attached logs (FS and windosws), a little analize and
> screenshot. Does anyone have the same issue?
>
>
>
> http://jira.freeswitch.org/browse/BKW-7
>
>
>
> regards arturo monroy
>
> amonroy at auronix.com
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> 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
>



-- 
Anthony Minessale II

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

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

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


From krice at freeswitch.org  Tue Aug  7 20:03:34 2012
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 7 Aug 2012 11:03:34 -0500
Subject: [Freeswitch-dev] Announcing FreeSWITCH 1.2
Message-ID: <CAPcNkdcr3--z9_Fi-r4Xp4wGqUFX5y7Ww5wjjBKq-Csv7mBz+A@mail.gmail.com>

The FreeSWITCH Team is Proud to announce FreeSWITCH 1.2.0!

Get your copy today at http://files.freeswitch.org/freeswitch-1.2.0.tar.bz2!

Going forward FreeSWITCH 1.2.x branch will be feature stable, but we will
continue to fix bugs and security issues. giving you a stable platform for
at least one year.

Grab it today!

The FreeSWITCH Team
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From jmesquita at gmail.com  Tue Aug  7 20:10:31 2012
From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Tue, 7 Aug 2012 13:10:31 -0300
Subject: [Freeswitch-dev] Announcing FreeSWITCH 1.2
In-Reply-To: <CAPcNkdcr3--z9_Fi-r4Xp4wGqUFX5y7Ww5wjjBKq-Csv7mBz+A@mail.gmail.com>
References: <CAPcNkdcr3--z9_Fi-r4Xp4wGqUFX5y7Ww5wjjBKq-Csv7mBz+A@mail.gmail.com>
Message-ID: <CAMYsERzr2soDwyKOG30rPYphvwETd3YdZpE_KW84pvoNvuZ_oA@mail.gmail.com>

HOOORRAY!!!!

If I don't have words to express what I feel with this milestone of the
project, I could only imagine the ones more deeply involved. All I can say
is cherish the moment. You all deserve it.

Special thanks to Tony for putting this all up. A piece of the world is in
debt with you.

Regards,

JM

On Tue, Aug 7, 2012 at 1:03 PM, Ken Rice <krice at freeswitch.org> wrote:

> The FreeSWITCH Team is Proud to announce FreeSWITCH 1.2.0!
>
> Get your copy today at
> http://files.freeswitch.org/freeswitch-1.2.0.tar.bz2 !
>
> Going forward FreeSWITCH 1.2.x branch will be feature stable, but we will
> continue to fix bugs and security issues. giving you a stable platform for
> at least one year.
>
> Grab it today!
>
> The FreeSWITCH Team
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> 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
>
>
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From anatoliy at kounitskiy.com  Tue Aug  7 20:48:08 2012
From: anatoliy at kounitskiy.com (Anatoliy Kounitskiy)
Date: Tue, 7 Aug 2012 19:48:08 +0300
Subject: [Freeswitch-dev] [Freeswitch-users] Announcing FreeSWITCH 1.2
In-Reply-To: <CAPcNkdcr3--z9_Fi-r4Xp4wGqUFX5y7Ww5wjjBKq-Csv7mBz+A@mail.gmail.com>
References: <CAPcNkdcr3--z9_Fi-r4Xp4wGqUFX5y7Ww5wjjBKq-Csv7mBz+A@mail.gmail.com>
Message-ID: <97D36ACD-601D-4D15-8BA0-E44D11BBC9A3@kounitskiy.com>

Time to start testing & planning migrations :D

Thanks!

On Aug 7, 2012, at 7:03 PM, Ken Rice <krice at freeswitch.org> wrote:

> The FreeSWITCH Team is Proud to announce FreeSWITCH 1.2.0!
> 
> Get your copy today at http://files.freeswitch.org/freeswitch-1.2.0.tar.bz2 ! 
> 
> Going forward FreeSWITCH 1.2.x branch will be feature stable, but we will continue to fix bugs and security issues. giving you a stable platform for at least one year.
> 
> Grab it today!
> 
> The FreeSWITCH Team 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From amonroy at auronix.com  Tue Aug  7 22:43:00 2012
From: amonroy at auronix.com (Arturo monroy)
Date: Tue, 7 Aug 2012 13:43:00 -0500
Subject: [Freeswitch-dev] FreeSwitch Crashes When "Sending Bye
	before	Blind_Transfer"
In-Reply-To: <CAKbxfG_9=j2F-xc952uJ=FR6F9LAvbaQ2VNfhYq9-e_czb0x=Q@mail.gmail.com>
References: <007f01cd7411$9ccd2ad0$d6678070$@auronix.com>
	<CAKbxfG_9=j2F-xc952uJ=FR6F9LAvbaQ2VNfhYq9-e_czb0x=Q@mail.gmail.com>
Message-ID: <000301cd74cc$78b79520$6a26bf60$@auronix.com>

Enterly my fault, thanks for reallocated

Saludos
Arturo Monroy

(55) 5371-1100 Ext 189
ASR Arturo Monroy
Auronix de Mexico
www.auronix.mx

-----Mensaje original-----
De: freeswitch-dev-bounces at lists.freeswitch.org
[mailto:freeswitch-dev-bounces at lists.freeswitch.org] En nombre de Anthony
Minessale
Enviado el: lunes, 06 de agosto de 2012 06:52 p.m.
Para: freeswitch-dev at lists.freeswitch.org
Asunto: Re: [Freeswitch-dev] FreeSwitch Crashes When "Sending Bye before
Blind_Transfer"

why did you file it under BKW and not FreeSWITCH that is not a real project
so nobody will see your post.


On Mon, Aug 6, 2012 at 3:25 PM, Arturo monroy <amonroy at auronix.com> wrote:
> Hey all!
>
>
>
>                 I've post a jira issue due FS crashes "appeared" after 
> sending bye before execute deflect command from Lua script, at jira 
> issue link I have attached logs (FS and windosws), a little analize 
> and screenshot. Does anyone have the same issue?
>
>
>
> http://jira.freeswitch.org/browse/BKW-7
>
>
>
> regards arturo monroy
>
> amonroy at auronix.com
>
>
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>  
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> 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
>



--
Anthony Minessale II

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

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

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

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




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

Join Us At ClueCon - Aug 7-9, 2012

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 amonroy at auronix.com  Tue Aug  7 22:46:15 2012
From: amonroy at auronix.com (Arturo monroy)
Date: Tue, 7 Aug 2012 13:46:15 -0500
Subject: [Freeswitch-dev] Announcing FreeSWITCH 1.2
In-Reply-To: <CAMYsERzr2soDwyKOG30rPYphvwETd3YdZpE_KW84pvoNvuZ_oA@mail.gmail.com>
References: <CAPcNkdcr3--z9_Fi-r4Xp4wGqUFX5y7Ww5wjjBKq-Csv7mBz+A@mail.gmail.com>
	<CAMYsERzr2soDwyKOG30rPYphvwETd3YdZpE_KW84pvoNvuZ_oA@mail.gmail.com>
Message-ID: <000401cd74cc$ecc7f450$c657dcf0$@auronix.com>

Congratulations!!

 

Saludos

Arturo Monroy

 

(55) 5371-1100 Ext 189

ASR Arturo Monroy

Auronix de Mexico

www.auronix.mx <http://www.auronix.mx/> 

 

De: freeswitch-dev-bounces at lists.freeswitch.org
[mailto:freeswitch-dev-bounces at lists.freeswitch.org] En nombre de Jo?o
Mesquita
Enviado el: martes, 07 de agosto de 2012 11:11 a.m.
Para: freeswitch-dev at lists.freeswitch.org
CC: FreeSWITCH Users Help
Asunto: Re: [Freeswitch-dev] Announcing FreeSWITCH 1.2

 

HOOORRAY!!!!

 

If I don't have words to express what I feel with this milestone of the
project, I could only imagine the ones more deeply involved. All I can say
is cherish the moment. You all deserve it.

 

Special thanks to Tony for putting this all up. A piece of the world is in
debt with you.

 

Regards,

 

JM

On Tue, Aug 7, 2012 at 1:03 PM, Ken Rice <krice at freeswitch.org> wrote:

The FreeSWITCH Team is Proud to announce FreeSWITCH 1.2.0!

 

Get your copy today at http://files.freeswitch.org/freeswitch-1.2.0.tar.bz2
! 

 

Going forward FreeSWITCH 1.2.x branch will be feature stable, but we will
continue to fix bugs and security issues. giving you a stable platform for
at least one year.

 

Grab it today!

 

The FreeSWITCH Team 


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




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

Join Us At ClueCon - Aug 7-9, 2012

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

 

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From snirkhey at gmail.com  Tue Aug  7 11:09:21 2012
From: snirkhey at gmail.com (saurabh nirkhey)
Date: Tue, 7 Aug 2012 12:39:21 +0530
Subject: [Freeswitch-dev] Creating a session without incoming or outgoing
	call
Message-ID: <CABuGx5MUsD0fbz42e3V1_AzEcOu8mSEXj08+Uk9zcCN3R9BUog@mail.gmail.com>

I am using mod_skel and writing my own module. My application needs me
to be able to create a session without an incoming call. My questions
are:

1. Is it possible to do this?
2. Do I need to inherit CoreSession or does an easier alternative exist?
3. Can I invoke my application from eventsocket without originating a
call at all?

Thanks for any inputs.

Regards,
Saurabh Nirkhey


From psycho.roxx at web.de  Tue Aug  7 23:00:49 2012
From: psycho.roxx at web.de (=?UTF-8?Q?=22Michael_K=C3=A4stle=22?=)
Date: Tue, 7 Aug 2012 21:00:49 +0200 (CEST)
Subject: [Freeswitch-dev] Information about FreeSWITCH
Message-ID: <trinity-5232a8de-0f8e-44ec-9625-b0203a841501-1344366049274@3capp-webde-bs41>

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From tomp at tomp.co.uk  Wed Aug  8 22:32:12 2012
From: tomp at tomp.co.uk (Tom Parrott)
Date: Wed, 08 Aug 2012 19:32:12 +0100
Subject: [Freeswitch-dev] 1.2.0 doesn't build on CentOS using RPM spec file
Message-ID: <5022B0AC.40007@tomp.co.uk>

Hi,

I have been building Freeswitch both git head and 1.2.rc2 on CentOS 5 
for some time.

However the stable tarball of 1.2.0 fails using the spec file supplied 
with the error:

libtool: link: ( cd ".libs" && rm -f "libcurl.la" && ln -s 
"../libcurl.la" "libcurl.la" )
make[3]: Leaving directory 
`/home/rpmbuild/BUILD/freeswitch-1.2.0/libs/curl/lib'
make[2]: Leaving directory 
`/home/rpmbuild/BUILD/freeswitch-1.2.0/libs/curl/lib'
Making all in src
make[2]: Entering directory 
`/home/rpmbuild/BUILD/freeswitch-1.2.0/libs/curl/src'
(CDPATH="${ZSH_VERSION+.}:" && cd .. && autoheader)
aclocal.m4:20: warning: this file was generated for autoconf 2.63.
You have another version of autoconf.  It may work, but is not 
guaranteed to.
If you have problems, you may need to regenerate the build system entirely.
To do so, use the procedure documented by the package, typically 
`autoreconf'.
configure.ac:65: error: Autoconf version 2.62 or higher is required
aclocal.m4:8467: AM_INIT_AUTOMAKE is expanded from...
configure.ac:65: the top level
autom4te: /usr/bin/m4 failed with exit status: 63
autoheader: /usr/bin/autom4te failed with exit status: 63
make[2]: *** [config.h.in] Error 63
make[2]: Leaving directory 
`/home/rpmbuild/BUILD/freeswitch-1.2.0/libs/curl/src'
make[1]: *** [all-recursive] Error 1
make[1]: Leaving directory `/home/rpmbuild/BUILD/freeswitch-1.2.0/libs/curl'
make: *** [libs/curl/lib/libcurl.la] Error 2
error: Bad exit status from /home/rpmbuild/tmp/rpm-tmp.30829 (%build)


RPM build errors:
     Bad exit status from /home/rpmbuild/tmp/rpm-tmp.30829 (%build)

Have the minimum requirements changed for Freeswitch?

Thanks
Tom


From tomp at tomp.co.uk  Wed Aug  8 23:23:38 2012
From: tomp at tomp.co.uk (Tom Parrott)
Date: Wed, 08 Aug 2012 20:23:38 +0100
Subject: [Freeswitch-dev] 1.2.0 doesn't build on CentOS using RPM spec
	file
In-Reply-To: <5022B0AC.40007@tomp.co.uk>
References: <5022B0AC.40007@tomp.co.uk>
Message-ID: <5022BCBA.2020207@tomp.co.uk>

I tried adding rebootstrap.sh to the build before running ./configure.

This fixed the first error, but later in the build I got another error:

     then mv -f ".deps/libfreeswitch_la-switch_apr.Tpo" 
".deps/libfreeswitch_la-switch_apr.Plo"; else rm -f 
".deps/libfreeswitch_la-switch_apr.Tpo"; exit 1; fi
/home/rpmbuild/BUILD/freeswitch-1.2.0/libtool: line 466: CDPATH: command 
not found
/home/rpmbuild/BUILD/freeswitch-1.2.0/libtool: line 1144: 
func_opt_split: command not found
libtool: Version mismatch error.  This is libtool 2.2.6b, but the
libtool: definition of this LT_INIT comes from an older release.
libtool: You should recreate aclocal.m4 with macros from libtool 2.2.6b
libtool: and run autoconf again.
make[1]: *** [libfreeswitch_la-switch_apr.lo] Error 1
make[1]: Leaving directory `/home/rpmbuild/BUILD/freeswitch-1.2.0'
make: *** [all] Error 2
error: Bad exit status from /home/rpmbuild/tmp/rpm-tmp.46063 (%build)


Thanks
Tom

On 08/08/12 19:32, Tom Parrott wrote:
> Hi,
>
> I have been building Freeswitch both git head and 1.2.rc2 on CentOS 5 
> for some time.
>
> However the stable tarball of 1.2.0 fails using the spec file supplied 
> with the error:
>
> libtool: link: ( cd ".libs" && rm -f "libcurl.la" && ln -s 
> "../libcurl.la" "libcurl.la" )
> make[3]: Leaving directory 
> `/home/rpmbuild/BUILD/freeswitch-1.2.0/libs/curl/lib'
> make[2]: Leaving directory 
> `/home/rpmbuild/BUILD/freeswitch-1.2.0/libs/curl/lib'
> Making all in src
> make[2]: Entering directory 
> `/home/rpmbuild/BUILD/freeswitch-1.2.0/libs/curl/src'
> (CDPATH="${ZSH_VERSION+.}:" && cd .. && autoheader)
> aclocal.m4:20: warning: this file was generated for autoconf 2.63.
> You have another version of autoconf.  It may work, but is not 
> guaranteed to.
> If you have problems, you may need to regenerate the build system 
> entirely.
> To do so, use the procedure documented by the package, typically 
> `autoreconf'.
> configure.ac:65: error: Autoconf version 2.62 or higher is required
> aclocal.m4:8467: AM_INIT_AUTOMAKE is expanded from...
> configure.ac:65: the top level
> autom4te: /usr/bin/m4 failed with exit status: 63
> autoheader: /usr/bin/autom4te failed with exit status: 63
> make[2]: *** [config.h.in] Error 63
> make[2]: Leaving directory 
> `/home/rpmbuild/BUILD/freeswitch-1.2.0/libs/curl/src'
> make[1]: *** [all-recursive] Error 1
> make[1]: Leaving directory 
> `/home/rpmbuild/BUILD/freeswitch-1.2.0/libs/curl'
> make: *** [libs/curl/lib/libcurl.la] Error 2
> error: Bad exit status from /home/rpmbuild/tmp/rpm-tmp.30829 (%build)
>
>
> RPM build errors:
>     Bad exit status from /home/rpmbuild/tmp/rpm-tmp.30829 (%build)
>
> Have the minimum requirements changed for Freeswitch?
>
> Thanks
> Tom



From krice at freeswitch.org  Wed Aug  8 23:52:06 2012
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 8 Aug 2012 14:52:06 -0500
Subject: [Freeswitch-dev] 1.2.0 doesn't build on CentOS using RPM spec
	file
In-Reply-To: <5022BCBA.2020207@tomp.co.uk>
References: <5022B0AC.40007@tomp.co.uk>
	<5022BCBA.2020207@tomp.co.uk>
Message-ID: <CAPcNkddAcz0_js+-JAbCqmxoc6cTy=8-ZmteWj=e_gsW211b4Q@mail.gmail.com>

This is a problem where your machine doesnt like the bootstrapped tarball
that you have... this is not a rpmbuild problem...

also this is an issue that should have been reported via jira.

We can not stress enough, please file all bugs via jira and add the
information requested so that we can know the version etc that you are
running

K

On Wed, Aug 8, 2012 at 2:23 PM, Tom Parrott <tomp at tomp.co.uk> wrote:

> I tried adding rebootstrap.sh to the build before running ./configure.
>
> This fixed the first error, but later in the build I got another error:
>
>      then mv -f ".deps/libfreeswitch_la-switch_apr.Tpo"
> ".deps/libfreeswitch_la-switch_apr.Plo"; else rm -f
> ".deps/libfreeswitch_la-switch_apr.Tpo"; exit 1; fi
> /home/rpmbuild/BUILD/freeswitch-1.2.0/libtool: line 466: CDPATH: command
> not found
> /home/rpmbuild/BUILD/freeswitch-1.2.0/libtool: line 1144:
> func_opt_split: command not found
> libtool: Version mismatch error.  This is libtool 2.2.6b, but the
> libtool: definition of this LT_INIT comes from an older release.
> libtool: You should recreate aclocal.m4 with macros from libtool 2.2.6b
> libtool: and run autoconf again.
> make[1]: *** [libfreeswitch_la-switch_apr.lo] Error 1
> make[1]: Leaving directory `/home/rpmbuild/BUILD/freeswitch-1.2.0'
> make: *** [all] Error 2
> error: Bad exit status from /home/rpmbuild/tmp/rpm-tmp.46063 (%build)
>
>
> Thanks
> Tom
>
> On 08/08/12 19:32, Tom Parrott wrote:
> > Hi,
> >
> > I have been building Freeswitch both git head and 1.2.rc2 on CentOS 5
> > for some time.
> >
> > However the stable tarball of 1.2.0 fails using the spec file supplied
> > with the error:
> >
> > libtool: link: ( cd ".libs" && rm -f "libcurl.la" && ln -s
> > "../libcurl.la" "libcurl.la" )
> > make[3]: Leaving directory
> > `/home/rpmbuild/BUILD/freeswitch-1.2.0/libs/curl/lib'
> > make[2]: Leaving directory
> > `/home/rpmbuild/BUILD/freeswitch-1.2.0/libs/curl/lib'
> > Making all in src
> > make[2]: Entering directory
> > `/home/rpmbuild/BUILD/freeswitch-1.2.0/libs/curl/src'
> > (CDPATH="${ZSH_VERSION+.}:" && cd .. && autoheader)
> > aclocal.m4:20: warning: this file was generated for autoconf 2.63.
> > You have another version of autoconf.  It may work, but is not
> > guaranteed to.
> > If you have problems, you may need to regenerate the build system
> > entirely.
> > To do so, use the procedure documented by the package, typically
> > `autoreconf'.
> > configure.ac:65: error: Autoconf version 2.62 or higher is required
> > aclocal.m4:8467: AM_INIT_AUTOMAKE is expanded from...
> > configure.ac:65: the top level
> > autom4te: /usr/bin/m4 failed with exit status: 63
> > autoheader: /usr/bin/autom4te failed with exit status: 63
> > make[2]: *** [config.h.in] Error 63
> > make[2]: Leaving directory
> > `/home/rpmbuild/BUILD/freeswitch-1.2.0/libs/curl/src'
> > make[1]: *** [all-recursive] Error 1
> > make[1]: Leaving directory
> > `/home/rpmbuild/BUILD/freeswitch-1.2.0/libs/curl'
> > make: *** [libs/curl/lib/libcurl.la] Error 2
> > error: Bad exit status from /home/rpmbuild/tmp/rpm-tmp.30829 (%build)
> >
> >
> > RPM build errors:
> >     Bad exit status from /home/rpmbuild/tmp/rpm-tmp.30829 (%build)
> >
> > Have the minimum requirements changed for Freeswitch?
> >
> > Thanks
> > Tom
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> 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
>
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From mahesoftengg at gmail.com  Thu Aug  9 08:30:35 2012
From: mahesoftengg at gmail.com (Mahesh R)
Date: Thu, 9 Aug 2012 10:00:35 +0530
Subject: [Freeswitch-dev] Doubt Regarding Interrupt a conference
Message-ID: <CANZX-88GkaOM4s19bvzq9TbR4R_LOcTQxHO11GmHo09wr=WbaA@mail.gmail.com>

Hi,

I have the following query. Let us assume that there are n users connected
from sip phone and m users who are connected from PSTN on the same
conference. If suppose say, one among the n or m users who are connected
via SIP phone or PSTN has a doubt/want to interrupt a conference, is there
any option like 'raise hand'. What I mean is, do we have any option of
dialing say some digit in the phone (PSTN) (when the conference is going
on) which in turn will notify other users that this user wants to raise
hand.


--
Regards,
Mahesh R
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From anthony.minessale at gmail.com  Thu Aug  9 18:43:26 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 9 Aug 2012 09:43:26 -0500
Subject: [Freeswitch-dev] Doubt Regarding Interrupt a conference
In-Reply-To: <CANZX-88GkaOM4s19bvzq9TbR4R_LOcTQxHO11GmHo09wr=WbaA@mail.gmail.com>
References: <CANZX-88GkaOM4s19bvzq9TbR4R_LOcTQxHO11GmHo09wr=WbaA@mail.gmail.com>
Message-ID: <CAKbxfG9i9eDM1WBLUvZMjCGJqPRBM71RrWP+S06=RsmiUQ9Xgw@mail.gmail.com>

not built into FS itself but the kind of functionality should be done
from your own custom app controlling the conference.


On Wed, Aug 8, 2012 at 11:30 PM, Mahesh R <mahesoftengg at gmail.com> wrote:
> Hi,
>
> I have the following query. Let us assume that there are n users connected
> from sip phone and m users who are connected from PSTN on the same
> conference. If suppose say, one among the n or m users who are connected via
> SIP phone or PSTN has a doubt/want to interrupt a conference, is there any
> option like 'raise hand'. What I mean is, do we have any option of dialing
> say some digit in the phone (PSTN) (when the conference is going on) which
> in turn will notify other users that this user wants to raise hand.
>
>
> --
> Regards,
> Mahesh R
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> 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
>



-- 
Anthony Minessale II

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

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

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


From tomp at tomp.co.uk  Fri Aug 10 22:01:38 2012
From: tomp at tomp.co.uk (Tom Parrott)
Date: Fri, 10 Aug 2012 19:01:38 +0100
Subject: [Freeswitch-dev] 1.2.0 doesn't build on CentOS using RPM spec
	file
In-Reply-To: <5022BCBA.2020207@tomp.co.uk>
References: <5022B0AC.40007@tomp.co.uk> <5022BCBA.2020207@tomp.co.uk>
Message-ID: <50254C82.5000401@tomp.co.uk>

Hi,

I gave up with the 1.2.0 tag branch, as it still wouldn't build when the 
1.2.0 tag was checked out from git.

I can report that the current git head builds perfectly though on CentOS 
5, so looks like the issue has been fixed already.

OK will log future stuff on Jira first.

Sometimes feels like I can't win, I got a telling off on the Jitsi SIP 
client mailing for opening a bug on Jira first, without discussing it on 
the mailing list!

Tom


On 08/08/12 20:23, Tom Parrott wrote:
> I tried adding rebootstrap.sh to the build before running ./configure.
>
> This fixed the first error, but later in the build I got another error:
>
>     then mv -f ".deps/libfreeswitch_la-switch_apr.Tpo" 
> ".deps/libfreeswitch_la-switch_apr.Plo"; else rm -f 
> ".deps/libfreeswitch_la-switch_apr.Tpo"; exit 1; fi
> /home/rpmbuild/BUILD/freeswitch-1.2.0/libtool: line 466: CDPATH: 
> command not found
> /home/rpmbuild/BUILD/freeswitch-1.2.0/libtool: line 1144: 
> func_opt_split: command not found
> libtool: Version mismatch error.  This is libtool 2.2.6b, but the
> libtool: definition of this LT_INIT comes from an older release.
> libtool: You should recreate aclocal.m4 with macros from libtool 2.2.6b
> libtool: and run autoconf again.
> make[1]: *** [libfreeswitch_la-switch_apr.lo] Error 1
> make[1]: Leaving directory `/home/rpmbuild/BUILD/freeswitch-1.2.0'
> make: *** [all] Error 2
> error: Bad exit status from /home/rpmbuild/tmp/rpm-tmp.46063 (%build)
>
>
> Thanks
> Tom
>
> On 08/08/12 19:32, Tom Parrott wrote:
>> Hi,
>>
>> I have been building Freeswitch both git head and 1.2.rc2 on CentOS 5 
>> for some time.
>>
>> However the stable tarball of 1.2.0 fails using the spec file 
>> supplied with the error:
>>
>> libtool: link: ( cd ".libs" && rm -f "libcurl.la" && ln -s 
>> "../libcurl.la" "libcurl.la" )
>> make[3]: Leaving directory 
>> `/home/rpmbuild/BUILD/freeswitch-1.2.0/libs/curl/lib'
>> make[2]: Leaving directory 
>> `/home/rpmbuild/BUILD/freeswitch-1.2.0/libs/curl/lib'
>> Making all in src
>> make[2]: Entering directory 
>> `/home/rpmbuild/BUILD/freeswitch-1.2.0/libs/curl/src'
>> (CDPATH="${ZSH_VERSION+.}:" && cd .. && autoheader)
>> aclocal.m4:20: warning: this file was generated for autoconf 2.63.
>> You have another version of autoconf.  It may work, but is not 
>> guaranteed to.
>> If you have problems, you may need to regenerate the build system 
>> entirely.
>> To do so, use the procedure documented by the package, typically 
>> `autoreconf'.
>> configure.ac:65: error: Autoconf version 2.62 or higher is required
>> aclocal.m4:8467: AM_INIT_AUTOMAKE is expanded from...
>> configure.ac:65: the top level
>> autom4te: /usr/bin/m4 failed with exit status: 63
>> autoheader: /usr/bin/autom4te failed with exit status: 63
>> make[2]: *** [config.h.in] Error 63
>> make[2]: Leaving directory 
>> `/home/rpmbuild/BUILD/freeswitch-1.2.0/libs/curl/src'
>> make[1]: *** [all-recursive] Error 1
>> make[1]: Leaving directory 
>> `/home/rpmbuild/BUILD/freeswitch-1.2.0/libs/curl'
>> make: *** [libs/curl/lib/libcurl.la] Error 2
>> error: Bad exit status from /home/rpmbuild/tmp/rpm-tmp.30829 (%build)
>>
>>
>> RPM build errors:
>>     Bad exit status from /home/rpmbuild/tmp/rpm-tmp.30829 (%build)
>>
>> Have the minimum requirements changed for Freeswitch?
>>
>> Thanks
>> Tom
>



From anthony.minessale at gmail.com  Fri Aug 10 23:20:42 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 10 Aug 2012 14:20:42 -0500
Subject: [Freeswitch-dev] 1.2.0 doesn't build on CentOS using RPM spec
	file
In-Reply-To: <50254C82.5000401@tomp.co.uk>
References: <5022B0AC.40007@tomp.co.uk> <5022BCBA.2020207@tomp.co.uk>
	<50254C82.5000401@tomp.co.uk>
Message-ID: <CAKbxfG-10JzEZQtSkciWxSqZCe43M=iBOoCvaJ_BDUu9O6aKfg@mail.gmail.com>

No sweat,

Or philosophy is its easier to tell one guy his Jira is not a bug than to
tell him cc:  2000 ppl that his email *is* a bug.

Ppl actually scan Jira looking to help more than they do the lists and
issues don't slip through the cracks on Jira.


On Aug 10, 2012 1:02 PM, "Tom Parrott" <tomp at tomp.co.uk> wrote:

> Hi,
>
> I gave up with the 1.2.0 tag branch, as it still wouldn't build when the
> 1.2.0 tag was checked out from git.
>
> I can report that the current git head builds perfectly though on CentOS
> 5, so looks like the issue has been fixed already.
>
> OK will log future stuff on Jira first.
>
> Sometimes feels like I can't win, I got a telling off on the Jitsi SIP
> client mailing for opening a bug on Jira first, without discussing it on
> the mailing list!
>
> Tom
>
>
> On 08/08/12 20:23, Tom Parrott wrote:
> > I tried adding rebootstrap.sh to the build before running ./configure.
> >
> > This fixed the first error, but later in the build I got another error:
> >
> >     then mv -f ".deps/libfreeswitch_la-switch_apr.Tpo"
> > ".deps/libfreeswitch_la-switch_apr.Plo"; else rm -f
> > ".deps/libfreeswitch_la-switch_apr.Tpo"; exit 1; fi
> > /home/rpmbuild/BUILD/freeswitch-1.2.0/libtool: line 466: CDPATH:
> > command not found
> > /home/rpmbuild/BUILD/freeswitch-1.2.0/libtool: line 1144:
> > func_opt_split: command not found
> > libtool: Version mismatch error.  This is libtool 2.2.6b, but the
> > libtool: definition of this LT_INIT comes from an older release.
> > libtool: You should recreate aclocal.m4 with macros from libtool 2.2.6b
> > libtool: and run autoconf again.
> > make[1]: *** [libfreeswitch_la-switch_apr.lo] Error 1
> > make[1]: Leaving directory `/home/rpmbuild/BUILD/freeswitch-1.2.0'
> > make: *** [all] Error 2
> > error: Bad exit status from /home/rpmbuild/tmp/rpm-tmp.46063 (%build)
> >
> >
> > Thanks
> > Tom
> >
> > On 08/08/12 19:32, Tom Parrott wrote:
> >> Hi,
> >>
> >> I have been building Freeswitch both git head and 1.2.rc2 on CentOS 5
> >> for some time.
> >>
> >> However the stable tarball of 1.2.0 fails using the spec file
> >> supplied with the error:
> >>
> >> libtool: link: ( cd ".libs" && rm -f "libcurl.la" && ln -s
> >> "../libcurl.la" "libcurl.la" )
> >> make[3]: Leaving directory
> >> `/home/rpmbuild/BUILD/freeswitch-1.2.0/libs/curl/lib'
> >> make[2]: Leaving directory
> >> `/home/rpmbuild/BUILD/freeswitch-1.2.0/libs/curl/lib'
> >> Making all in src
> >> make[2]: Entering directory
> >> `/home/rpmbuild/BUILD/freeswitch-1.2.0/libs/curl/src'
> >> (CDPATH="${ZSH_VERSION+.}:" && cd .. && autoheader)
> >> aclocal.m4:20: warning: this file was generated for autoconf 2.63.
> >> You have another version of autoconf.  It may work, but is not
> >> guaranteed to.
> >> If you have problems, you may need to regenerate the build system
> >> entirely.
> >> To do so, use the procedure documented by the package, typically
> >> `autoreconf'.
> >> configure.ac:65: error: Autoconf version 2.62 or higher is required
> >> aclocal.m4:8467: AM_INIT_AUTOMAKE is expanded from...
> >> configure.ac:65: the top level
> >> autom4te: /usr/bin/m4 failed with exit status: 63
> >> autoheader: /usr/bin/autom4te failed with exit status: 63
> >> make[2]: *** [config.h.in] Error 63
> >> make[2]: Leaving directory
> >> `/home/rpmbuild/BUILD/freeswitch-1.2.0/libs/curl/src'
> >> make[1]: *** [all-recursive] Error 1
> >> make[1]: Leaving directory
> >> `/home/rpmbuild/BUILD/freeswitch-1.2.0/libs/curl'
> >> make: *** [libs/curl/lib/libcurl.la] Error 2
> >> error: Bad exit status from /home/rpmbuild/tmp/rpm-tmp.30829 (%build)
> >>
> >>
> >> RPM build errors:
> >>     Bad exit status from /home/rpmbuild/tmp/rpm-tmp.30829 (%build)
> >>
> >> Have the minimum requirements changed for Freeswitch?
> >>
> >> Thanks
> >> Tom
> >
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> 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
>
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From krice at freeswitch.org  Sat Aug 11 01:23:17 2012
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 10 Aug 2012 16:23:17 -0500
Subject: [Freeswitch-dev] 1.2.0 doesn't build on CentOS using RPM spec
	file
In-Reply-To: <50254C82.5000401@tomp.co.uk>
References: <5022B0AC.40007@tomp.co.uk> <5022BCBA.2020207@tomp.co.uk>
	<50254C82.5000401@tomp.co.uk>
Message-ID: <3DB87FE3-A714-4B27-84C9-A867807117FC@freeswitch.org>

the 1.2 branch builds on centos 6 if you start with a clean tree if you already have cruft from master/HEAD you will see issues... also keep in mind that many of us are still travelling home from ClueCon so that means internet connectivity is iffy at best

Ken
Sent from my iPad

On Aug 10, 2012, at 1:01 PM, Tom Parrott <tomp at tomp.co.uk> wrote:

> Hi,
> 
> I gave up with the 1.2.0 tag branch, as it still wouldn't build when the 
> 1.2.0 tag was checked out from git.
> 
> I can report that the current git head builds perfectly though on CentOS 
> 5, so looks like the issue has been fixed already.
> 
> OK will log future stuff on Jira first.
> 
> Sometimes feels like I can't win, I got a telling off on the Jitsi SIP 
> client mailing for opening a bug on Jira first, without discussing it on 
> the mailing list!
> 
> Tom
> 
> 
> On 08/08/12 20:23, Tom Parrott wrote:
>> I tried adding rebootstrap.sh to the build before running ./configure.
>> 
>> This fixed the first error, but later in the build I got another error:
>> 
>>    then mv -f ".deps/libfreeswitch_la-switch_apr.Tpo" 
>> ".deps/libfreeswitch_la-switch_apr.Plo"; else rm -f 
>> ".deps/libfreeswitch_la-switch_apr.Tpo"; exit 1; fi
>> /home/rpmbuild/BUILD/freeswitch-1.2.0/libtool: line 466: CDPATH: 
>> command not found
>> /home/rpmbuild/BUILD/freeswitch-1.2.0/libtool: line 1144: 
>> func_opt_split: command not found
>> libtool: Version mismatch error.  This is libtool 2.2.6b, but the
>> libtool: definition of this LT_INIT comes from an older release.
>> libtool: You should recreate aclocal.m4 with macros from libtool 2.2.6b
>> libtool: and run autoconf again.
>> make[1]: *** [libfreeswitch_la-switch_apr.lo] Error 1
>> make[1]: Leaving directory `/home/rpmbuild/BUILD/freeswitch-1.2.0'
>> make: *** [all] Error 2
>> error: Bad exit status from /home/rpmbuild/tmp/rpm-tmp.46063 (%build)
>> 
>> 
>> Thanks
>> Tom
>> 
>> On 08/08/12 19:32, Tom Parrott wrote:
>>> Hi,
>>> 
>>> I have been building Freeswitch both git head and 1.2.rc2 on CentOS 5 
>>> for some time.
>>> 
>>> However the stable tarball of 1.2.0 fails using the spec file 
>>> supplied with the error:
>>> 
>>> libtool: link: ( cd ".libs" && rm -f "libcurl.la" && ln -s 
>>> "../libcurl.la" "libcurl.la" )
>>> make[3]: Leaving directory 
>>> `/home/rpmbuild/BUILD/freeswitch-1.2.0/libs/curl/lib'
>>> make[2]: Leaving directory 
>>> `/home/rpmbuild/BUILD/freeswitch-1.2.0/libs/curl/lib'
>>> Making all in src
>>> make[2]: Entering directory 
>>> `/home/rpmbuild/BUILD/freeswitch-1.2.0/libs/curl/src'
>>> (CDPATH="${ZSH_VERSION+.}:" && cd .. && autoheader)
>>> aclocal.m4:20: warning: this file was generated for autoconf 2.63.
>>> You have another version of autoconf.  It may work, but is not 
>>> guaranteed to.
>>> If you have problems, you may need to regenerate the build system 
>>> entirely.
>>> To do so, use the procedure documented by the package, typically 
>>> `autoreconf'.
>>> configure.ac:65: error: Autoconf version 2.62 or higher is required
>>> aclocal.m4:8467: AM_INIT_AUTOMAKE is expanded from...
>>> configure.ac:65: the top level
>>> autom4te: /usr/bin/m4 failed with exit status: 63
>>> autoheader: /usr/bin/autom4te failed with exit status: 63
>>> make[2]: *** [config.h.in] Error 63
>>> make[2]: Leaving directory 
>>> `/home/rpmbuild/BUILD/freeswitch-1.2.0/libs/curl/src'
>>> make[1]: *** [all-recursive] Error 1
>>> make[1]: Leaving directory 
>>> `/home/rpmbuild/BUILD/freeswitch-1.2.0/libs/curl'
>>> make: *** [libs/curl/lib/libcurl.la] Error 2
>>> error: Bad exit status from /home/rpmbuild/tmp/rpm-tmp.30829 (%build)
>>> 
>>> 
>>> RPM build errors:
>>>    Bad exit status from /home/rpmbuild/tmp/rpm-tmp.30829 (%build)
>>> 
>>> Have the minimum requirements changed for Freeswitch?
>>> 
>>> Thanks
>>> Tom
>> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> Join Us At ClueCon - Aug 7-9, 2012
> 
> 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 tomp at tomp.co.uk  Sat Aug 11 14:45:28 2012
From: tomp at tomp.co.uk (Tom Parrott)
Date: Sat, 11 Aug 2012 11:45:28 +0100
Subject: [Freeswitch-dev] 1.2.0 doesn't build on CentOS using RPM spec
	file
In-Reply-To: <50254C82.5000401@tomp.co.uk>
References: <5022B0AC.40007@tomp.co.uk> <5022BCBA.2020207@tomp.co.uk>
	<50254C82.5000401@tomp.co.uk>
Message-ID: <502637C8.9010909@tomp.co.uk>

No problem, I have opened up a bug report on Jira for completeness.

Hope the conference went well! Any chance of a Cluecon UK? :)

Tom

On 10/08/12 19:01, Tom Parrott wrote:
> Hi,
>
> I gave up with the 1.2.0 tag branch, as it still wouldn't build when 
> the 1.2.0 tag was checked out from git.
>
> I can report that the current git head builds perfectly though on 
> CentOS 5, so looks like the issue has been fixed already.
>
> OK will log future stuff on Jira first.
>
> Sometimes feels like I can't win, I got a telling off on the Jitsi SIP 
> client mailing for opening a bug on Jira first, without discussing it 
> on the mailing list!
>
> Tom
>
>
> On 08/08/12 20:23, Tom Parrott wrote:
>> I tried adding rebootstrap.sh to the build before running ./configure.
>>
>> This fixed the first error, but later in the build I got another error:
>>
>>     then mv -f ".deps/libfreeswitch_la-switch_apr.Tpo" 
>> ".deps/libfreeswitch_la-switch_apr.Plo"; else rm -f 
>> ".deps/libfreeswitch_la-switch_apr.Tpo"; exit 1; fi
>> /home/rpmbuild/BUILD/freeswitch-1.2.0/libtool: line 466: CDPATH: 
>> command not found
>> /home/rpmbuild/BUILD/freeswitch-1.2.0/libtool: line 1144: 
>> func_opt_split: command not found
>> libtool: Version mismatch error.  This is libtool 2.2.6b, but the
>> libtool: definition of this LT_INIT comes from an older release.
>> libtool: You should recreate aclocal.m4 with macros from libtool 2.2.6b
>> libtool: and run autoconf again.
>> make[1]: *** [libfreeswitch_la-switch_apr.lo] Error 1
>> make[1]: Leaving directory `/home/rpmbuild/BUILD/freeswitch-1.2.0'
>> make: *** [all] Error 2
>> error: Bad exit status from /home/rpmbuild/tmp/rpm-tmp.46063 (%build)
>>
>>
>> Thanks
>> Tom
>>
>> On 08/08/12 19:32, Tom Parrott wrote:
>>> Hi,
>>>
>>> I have been building Freeswitch both git head and 1.2.rc2 on CentOS 
>>> 5 for some time.
>>>
>>> However the stable tarball of 1.2.0 fails using the spec file 
>>> supplied with the error:
>>>
>>> libtool: link: ( cd ".libs" && rm -f "libcurl.la" && ln -s 
>>> "../libcurl.la" "libcurl.la" )
>>> make[3]: Leaving directory 
>>> `/home/rpmbuild/BUILD/freeswitch-1.2.0/libs/curl/lib'
>>> make[2]: Leaving directory 
>>> `/home/rpmbuild/BUILD/freeswitch-1.2.0/libs/curl/lib'
>>> Making all in src
>>> make[2]: Entering directory 
>>> `/home/rpmbuild/BUILD/freeswitch-1.2.0/libs/curl/src'
>>> (CDPATH="${ZSH_VERSION+.}:" && cd .. && autoheader)
>>> aclocal.m4:20: warning: this file was generated for autoconf 2.63.
>>> You have another version of autoconf.  It may work, but is not 
>>> guaranteed to.
>>> If you have problems, you may need to regenerate the build system 
>>> entirely.
>>> To do so, use the procedure documented by the package, typically 
>>> `autoreconf'.
>>> configure.ac:65: error: Autoconf version 2.62 or higher is required
>>> aclocal.m4:8467: AM_INIT_AUTOMAKE is expanded from...
>>> configure.ac:65: the top level
>>> autom4te: /usr/bin/m4 failed with exit status: 63
>>> autoheader: /usr/bin/autom4te failed with exit status: 63
>>> make[2]: *** [config.h.in] Error 63
>>> make[2]: Leaving directory 
>>> `/home/rpmbuild/BUILD/freeswitch-1.2.0/libs/curl/src'
>>> make[1]: *** [all-recursive] Error 1
>>> make[1]: Leaving directory 
>>> `/home/rpmbuild/BUILD/freeswitch-1.2.0/libs/curl'
>>> make: *** [libs/curl/lib/libcurl.la] Error 2
>>> error: Bad exit status from /home/rpmbuild/tmp/rpm-tmp.30829 (%build)
>>>
>>>
>>> RPM build errors:
>>>     Bad exit status from /home/rpmbuild/tmp/rpm-tmp.30829 (%build)
>>>
>>> Have the minimum requirements changed for Freeswitch?
>>>
>>> Thanks
>>> Tom
>>
>



From tomp at tomp.co.uk  Sat Aug 11 18:02:03 2012
From: tomp at tomp.co.uk (Tom Parrott)
Date: Sat, 11 Aug 2012 15:02:03 +0100
Subject: [Freeswitch-dev] Level3 and 25s failover
Message-ID: <502665DB.1060202@tomp.co.uk>

Hi,

The company I work for get toll-free numbers from Level3 via 
Bandwidth.com which are terminated on several Freeswitch servers using 
DNS SRV records.

Annoyingly Level3 have a 'feature' where they will failover a call to a 
different server if the call is not answered in 25 seconds.

In the UK, the provider we use has a similar failover setting, but as 
soon as early media is sent the fail-over mechanism is stopped and you 
can ring for as long as you need to.

I have been told by Level3 that it is a known issue, but not something 
they can/will change, and that calls should be answered within 25 seconds.

I tend to agree, however I forward calls to our customer's PSTN numbers, 
and they do not always answer within that time.

This causes duplicate call records when the call fails over. For a call 
analytics company, this causes serious problems in statistics.

Does anyone have any experience with Level3 and this strange behaviour?

Thanks
Tom





From anthony.minessale at gmail.com  Sat Aug 11 23:16:54 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 11 Aug 2012 14:16:54 -0500
Subject: [Freeswitch-dev] Level3 and 25s failover
In-Reply-To: <502665DB.1060202@tomp.co.uk>
References: <502665DB.1060202@tomp.co.uk>
Message-ID: <CAKbxfG_mSTMYBSqZm-3a6rKuhpcxLkBNJpdc5yLcut15GJE8RA@mail.gmail.com>

That seems like a strange rule.
I would think that if you sent any form of progress indication 18x that
they should wait a lot longer than 25
On Aug 11, 2012 9:04 AM, "Tom Parrott" <tomp at tomp.co.uk> wrote:

> Hi,
>
> The company I work for get toll-free numbers from Level3 via
> Bandwidth.com which are terminated on several Freeswitch servers using
> DNS SRV records.
>
> Annoyingly Level3 have a 'feature' where they will failover a call to a
> different server if the call is not answered in 25 seconds.
>
> In the UK, the provider we use has a similar failover setting, but as
> soon as early media is sent the fail-over mechanism is stopped and you
> can ring for as long as you need to.
>
> I have been told by Level3 that it is a known issue, but not something
> they can/will change, and that calls should be answered within 25 seconds.
>
> I tend to agree, however I forward calls to our customer's PSTN numbers,
> and they do not always answer within that time.
>
> This causes duplicate call records when the call fails over. For a call
> analytics company, this causes serious problems in statistics.
>
> Does anyone have any experience with Level3 and this strange behaviour?
>
> Thanks
> Tom
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> Join Us At ClueCon - Aug 7-9, 2012
>
> 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
>
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From krice at freeswitch.org  Tue Aug 14 11:13:21 2012
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 14 Aug 2012 02:13:21 -0500
Subject: [Freeswitch-dev] FreeSWITCH 1.2.1 has been rolled...
Message-ID: <CC4F64C1.3C00B%krice@freeswitch.org>

FreeSWITCH 1.2.1 is Here!!!

Tarball is on files.freeswitch.org as usual.

Also Debian users, check out the debian repo on
http://files.freeswitch.org/repo  More info on this coming soon, currently
this continues nightly builds from the dev branch but will soon also contain
a stable repo... 

As usual, any issues please report them via jira.freeswitch.org and please
tag them with the correct version!


K 
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From peter.olsson at visionutveckling.se  Wed Aug 15 16:50:09 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 15 Aug 2012 12:50:09 +0000
Subject: [Freeswitch-dev] Rare audio "clicks" inside mod_conference
Message-ID: <1FFF97C269757C458224B7C895F35F151455C9@cantor.std.visionutv.se>

I'm not really sure where to ask about this, but I'll give it a try on the dev-list for starters..

I sometimes (very rarely, but often enough to want to find the cause) get audio "clicks" in the audio stream on devices connected to a FS conference. I have samples from wireshark showing the problem. In L16 format it seems there are a few bytes set to 32767 (max int16 value), that is causing the actual audio.

The strange thing is that when this occurs, there is actually no one that sends this audio (but all members except one will get this audio, so it has been put into the buffer by the member without the click), so I'm starting to think if this is a generated packet inside FS, that is not initialized correctly, or if the decoding from PCMA/PCMU to L16 fails for some reason.

I've traced it down so far that I know how to handle it (at least work around it), but I don't know how to reproduce it, and I don't know the real cause. With the help of lots of debugging I've seen that this occurs when a partial frame is read from a member (however, according to wireshark, this partial frame was never sent over the network). We have 8khz conferences, and the connected members use PCMA and/or PCMU, so normally when audio from a member is appended to the audio buffer, the length of the data is 320 bytes (I guess this is L16). When the click occurs, I've seen that the datasize is only 160 bytes instead, which would more indicate a undecoded PCMA/PCMU frame. And as I said, there is no packet with half the payload size anywhere in wireshark, so I'm not really sure where this is coming from. I've looked through most of the code in switch_rtp.c and mod_sofia.c (that's related to RTP), but I can't really find any good reason.

Right now I've added a check (if read_frame->datalen != member->read_impl.encoded_bytes_per_packet), and when using this it works as expected (at least so far - as I said, it happens rarely), but I guess I might miss a frame of audio. I will try to look into this further, but if anyone have any suggestions I'm open for ideas :)

/Peter
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From peter.olsson at visionutveckling.se  Wed Aug 15 16:57:40 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 15 Aug 2012 12:57:40 +0000
Subject: [Freeswitch-dev] Rare audio "clicks" inside mod_conference
Message-ID: <1FFF97C269757C458224B7C895F35F151456BB@cantor.std.visionutv.se>

Before anyone mentions it (since I'm usually one of those who does), I will submit this to Jira, but I would appreciate some more input before doing so :)

/Peter

Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] F?r Peter Olsson
Skickat: den 15 augusti 2012 14:50
Till: freeswitch-dev at lists.freeswitch.org
?mne: [Freeswitch-dev] Rare audio "clicks" inside mod_conference

I'm not really sure where to ask about this, but I'll give it a try on the dev-list for starters..

I sometimes (very rarely, but often enough to want to find the cause) get audio "clicks" in the audio stream on devices connected to a FS conference. I have samples from wireshark showing the problem. In L16 format it seems there are a few bytes set to 32767 (max int16 value), that is causing the actual audio.

The strange thing is that when this occurs, there is actually no one that sends this audio (but all members except one will get this audio, so it has been put into the buffer by the member without the click), so I'm starting to think if this is a generated packet inside FS, that is not initialized correctly, or if the decoding from PCMA/PCMU to L16 fails for some reason.

I've traced it down so far that I know how to handle it (at least work around it), but I don't know how to reproduce it, and I don't know the real cause. With the help of lots of debugging I've seen that this occurs when a partial frame is read from a member (however, according to wireshark, this partial frame was never sent over the network). We have 8khz conferences, and the connected members use PCMA and/or PCMU, so normally when audio from a member is appended to the audio buffer, the length of the data is 320 bytes (I guess this is L16). When the click occurs, I've seen that the datasize is only 160 bytes instead, which would more indicate a undecoded PCMA/PCMU frame. And as I said, there is no packet with half the payload size anywhere in wireshark, so I'm not really sure where this is coming from. I've looked through most of the code in switch_rtp.c and mod_sofia.c (that's related to RTP), but I can't really find any good reason.

Right now I've added a check (if read_frame->datalen != member->read_impl.encoded_bytes_per_packet), and when using this it works as expected (at least so far - as I said, it happens rarely), but I guess I might miss a frame of audio. I will try to look into this further, but if anyone have any suggestions I'm open for ideas :)

/Peter
!DSPAM:502b98ee32764854613503!
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From krice at freeswitch.org  Wed Aug 15 19:31:31 2012
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 15 Aug 2012 10:31:31 -0500
Subject: [Freeswitch-dev] Weekly FreeSWITCH Conference Call
Message-ID: <CC512B03.3C07B%krice@freeswitch.org>

Hello Gang!

This weeks conf call will be starting soon...

The Agenda Page for today:
http://wiki.freeswitch.org/wiki/FS_weekly_2012_08_15

I will be talking about FreeSWITCH 1.2, how to get it and whats news.

We?ll follow this up with community discussion!

Join us!

Ken
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From anthony.minessale at gmail.com  Wed Aug 15 20:44:46 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 15 Aug 2012 11:44:46 -0500
Subject: [Freeswitch-dev] Rare audio "clicks" inside mod_conference
In-Reply-To: <1FFF97C269757C458224B7C895F35F151456BB@cantor.std.visionutv.se>
References: <1FFF97C269757C458224B7C895F35F151456BB@cantor.std.visionutv.se>
Message-ID: <CAKbxfG-KwRuGZuGihqOHYO9HcgEYyhktDu397xS-un==LpicqQ@mail.gmail.com>

Add some debugging to the places in switch_core_io.c that do memset,
maybe its from that.


On Wed, Aug 15, 2012 at 7:57 AM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:
> Before anyone mentions it (since I?m usually one of those who does), I will
> submit this to Jira, but I would appreciate some more input before doing so
> :)
>
>
>
> /Peter
>
>
>
> Fr?n: freeswitch-dev-bounces at lists.freeswitch.org
> [mailto:freeswitch-dev-bounces at lists.freeswitch.org] F?r Peter Olsson
> Skickat: den 15 augusti 2012 14:50
> Till: freeswitch-dev at lists.freeswitch.org
> ?mne: [Freeswitch-dev] Rare audio "clicks" inside mod_conference
>
>
>
> I?m not really sure where to ask about this, but I?ll give it a try on the
> dev-list for starters..
>
>
>
> I sometimes (very rarely, but often enough to want to find the cause) get
> audio ?clicks? in the audio stream on devices connected to a FS conference.
> I have samples from wireshark showing the problem. In L16 format it seems
> there are a few bytes set to 32767 (max int16 value), that is causing the
> actual audio.
>
>
>
> The strange thing is that when this occurs, there is actually no one that
> sends this audio (but all members except one will get this audio, so it has
> been put into the buffer by the member without the click), so I?m starting
> to think if this is a generated packet inside FS, that is not initialized
> correctly, or if the decoding from PCMA/PCMU to L16 fails for some reason.
>
>
>
> I?ve traced it down so far that I know how to handle it (at least work
> around it), but I don?t know how to reproduce it, and I don?t know the real
> cause. With the help of lots of debugging I?ve seen that this occurs when a
> partial frame is read from a member (however, according to wireshark, this
> partial frame was never sent over the network). We have 8khz conferences,
> and the connected members use PCMA and/or PCMU, so normally when audio from
> a member is appended to the audio buffer, the length of the data is 320
> bytes (I guess this is L16). When the click occurs, I?ve seen that the
> datasize is only 160 bytes instead, which would more indicate a undecoded
> PCMA/PCMU frame. And as I said, there is no packet with half the payload
> size anywhere in wireshark, so I?m not really sure where this is coming
> from. I?ve looked through most of the code in switch_rtp.c and mod_sofia.c
> (that?s related to RTP), but I can?t really find any good reason.
>
>
>
> Right now I?ve added a check (if read_frame->datalen !=
> member->read_impl.encoded_bytes_per_packet), and when using this it works as
> expected (at least so far ? as I said, it happens rarely), but I guess I
> might miss a frame of audio. I will try to look into this further, but if
> anyone have any suggestions I?m open for ideas :)
>
>
>
> /Peter
>
> !DSPAM:502b98ee32764854613503!
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> 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
>



-- 
Anthony Minessale II

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

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

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


From piyush.singhai at eng.knowlarity.com  Thu Aug 16 14:58:02 2012
From: piyush.singhai at eng.knowlarity.com (piyush singhai)
Date: Thu, 16 Aug 2012 16:28:02 +0530
Subject: [Freeswitch-dev] Mod_Callcenter queue and hashtable
Message-ID: <CAHHSTJ55Yx=G0z+_4mFHARV2s1yEgHvvLNyOOpYNq=mb+Wk9Nw@mail.gmail.com>

Hello,

I am trying to change the code in mod_callcenter. I want to implement queue
add functionality in the code currently it is not available.

i followed this link http://jira.freeswitch.org/browse/FS-3338 to add a
patch in the mod_callcenter.c file and add the queue also in the database.

I have some questions while following the code. Please correct me if my
understanding is wrong.
1. There is no database table for queue that is created in the xml and
after applying the patch also. after following the code i got that it is
stored in the hashtable. So why i can not see in the database and why it is
storing in the hashtable.? How mod_Callcenter storing queue information in
the freeswitch.

2. Is there  any other way to add queue at runtime.??

3. what is stored in the members table?

Please help me out.
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From stephen at picardogroup.com  Fri Aug 17 07:11:11 2012
From: stephen at picardogroup.com (stephen at picardogroup.com)
Date: Thu, 16 Aug 2012 20:11:11 -0700
Subject: [Freeswitch-dev] Audio Delay
Message-ID: <20120816201111.0e1bd4d5c5064b420440751b21b10e46.de292f4b50.wbe@email13.secureserver.net>

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From peter.olsson at visionutveckling.se  Fri Aug 17 10:34:47 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 17 Aug 2012 06:34:47 +0000
Subject: [Freeswitch-dev] Audio Delay
Message-ID: <1FFF97C269757C458224B7C895F35F15146D8E@cantor.std.visionutv.se>

Nope... :) Are you on real hardware, or a virtual environment?

Try issuing ?timer_test? in the console or the fs_cli, check if the results look ok ? usually this is a timing problem, but there are lots of other possibilities ? network, hardware problems etc.

/Peter

Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] F?r stephen at picardogroup.com
Skickat: den 17 augusti 2012 05:11
Till: freeswitch-dev at lists.freeswitch.org
?mne: [Freeswitch-dev] Audio Delay

Cent OS 5.4, Intel I7,  FS 1.2 Latest release , sangoma 101d card with latest driver (wanpipe) PRI circuit.

When two people in conference, we are experiencing increased audio delay as call goes longer.
Example in a 30 min time period after just 4-5 mins, the delay gets longer and longer reaching almost 1 minute after 10 mins or so. Anyone else experienced this problem?
!DSPAM:502ddf1b32761645758032!
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From peter.olsson at visionutveckling.se  Fri Aug 17 13:51:49 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 17 Aug 2012 09:51:49 +0000
Subject: [Freeswitch-dev] Rare audio "clicks" inside mod_conference
Message-ID: <1FFF97C269757C458224B7C895F35F151471D0@cantor.std.visionutv.se>

Thanks for the tip - it got me thinking in the right direction. I think I've found it now, and I'm running my changed code in production for some testing. If everything looks good I'll post a patch on Jira after the weekend.

/Peter

-----Ursprungligt meddelande-----
Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] F?r Anthony Minessale
Skickat: den 15 augusti 2012 18:45
Till: freeswitch-dev at lists.freeswitch.org
?mne: Re: [Freeswitch-dev] Rare audio "clicks" inside mod_conference

Add some debugging to the places in switch_core_io.c that do memset, maybe its from that.


On Wed, Aug 15, 2012 at 7:57 AM, Peter Olsson <peter.olsson at visionutveckling.se> wrote:
> Before anyone mentions it (since I'm usually one of those who does), I 
> will submit this to Jira, but I would appreciate some more input 
> before doing so
> :)
>
>
>
> /Peter
>
>
>
> Fr?n: freeswitch-dev-bounces at lists.freeswitch.org
> [mailto:freeswitch-dev-bounces at lists.freeswitch.org] F?r Peter Olsson
> Skickat: den 15 augusti 2012 14:50
> Till: freeswitch-dev at lists.freeswitch.org
> ?mne: [Freeswitch-dev] Rare audio "clicks" inside mod_conference
>
>
>
> I'm not really sure where to ask about this, but I'll give it a try on 
> the dev-list for starters..
>
>
>
> I sometimes (very rarely, but often enough to want to find the cause) 
> get audio "clicks" in the audio stream on devices connected to a FS conference.
> I have samples from wireshark showing the problem. In L16 format it 
> seems there are a few bytes set to 32767 (max int16 value), that is 
> causing the actual audio.
>
>
>
> The strange thing is that when this occurs, there is actually no one 
> that sends this audio (but all members except one will get this audio, 
> so it has been put into the buffer by the member without the click), 
> so I'm starting to think if this is a generated packet inside FS, that 
> is not initialized correctly, or if the decoding from PCMA/PCMU to L16 fails for some reason.
>
>
>
> I've traced it down so far that I know how to handle it (at least work 
> around it), but I don't know how to reproduce it, and I don't know the 
> real cause. With the help of lots of debugging I've seen that this 
> occurs when a partial frame is read from a member (however, according 
> to wireshark, this partial frame was never sent over the network). We 
> have 8khz conferences, and the connected members use PCMA and/or PCMU, 
> so normally when audio from a member is appended to the audio buffer, 
> the length of the data is 320 bytes (I guess this is L16). When the 
> click occurs, I've seen that the datasize is only 160 bytes instead, 
> which would more indicate a undecoded PCMA/PCMU frame. And as I said, 
> there is no packet with half the payload size anywhere in wireshark, 
> so I'm not really sure where this is coming from. I've looked through 
> most of the code in switch_rtp.c and mod_sofia.c (that's related to RTP), but I can't really find any good reason.
>
>
>
> Right now I've added a check (if read_frame->datalen !=
> member->read_impl.encoded_bytes_per_packet), and when using this it 
> member->works as
> expected (at least so far - as I said, it happens rarely), but I guess 
> I might miss a frame of audio. I will try to look into this further, 
> but if anyone have any suggestions I'm open for ideas :)
>
>
>
> /Peter
>
> 
>
>
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>  
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> 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
>



--
Anthony Minessale II

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

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

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

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

 

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

FreeSWITCH-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

!DSPAM:502bcf8532761778320602!



From stephen at picardogroup.com  Fri Aug 17 17:32:49 2012
From: stephen at picardogroup.com (stephen at picardogroup.com)
Date: Fri, 17 Aug 2012 06:32:49 -0700
Subject: [Freeswitch-dev] Audio Delay
Message-ID: <20120817063249.0e1bd4d5c5064b420440751b21b10e46.0ef5862238.wbe@email13.secureserver.net>

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From stephen at picardogroup.com  Sun Aug 19 02:44:51 2012
From: stephen at picardogroup.com (stephen at picardogroup.com)
Date: Sat, 18 Aug 2012 15:44:51 -0700
Subject: [Freeswitch-dev] Forcing FS to use Sangoma timer
Message-ID: <20120818154451.0e1bd4d5c5064b420440751b21b10e46.d040952c49.wbe@email13.secureserver.net>

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From stephen at picardogroup.com  Sun Aug 19 02:48:32 2012
From: stephen at picardogroup.com (stephen at picardogroup.com)
Date: Sat, 18 Aug 2012 15:48:32 -0700
Subject: [Freeswitch-dev] forcing FS to use sangoma timer
Message-ID: <20120818154832.0e1bd4d5c5064b420440751b21b10e46.8d4421edcd.wbe@email13.secureserver.net>

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From peter.olsson at visionutveckling.se  Mon Aug 20 20:13:38 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 20 Aug 2012 16:13:38 +0000
Subject: [Freeswitch-dev] Forcing FS to use Sangoma timer
In-Reply-To: <20120818154451.0e1bd4d5c5064b420440751b21b10e46.d040952c49.wbe@email13.secureserver.net>
References: <20120818154451.0e1bd4d5c5064b420440751b21b10e46.d040952c49.wbe@email13.secureserver.net>
Message-ID: <29D5CD49-E078-4FAE-A62C-5D9A999C811C@visionutveckling.se>

Stephen, as I mentoned to you earlier, there is no such thing as a sangoma timer module in FS, so this is not possible.

/Peter

20 aug 2012 kl. 17:17 skrev "stephen at picardogroup.com<mailto:stephen at picardogroup.com>" <stephen at picardogroup.com<mailto:stephen at picardogroup.com>>:

Have a FS application that is experiencing audio delay that increases as conference goes on longer. CENT OS 5, FS 1.2, 101D

Timer test results: 2012-08-18 17:47:25.838233 [CONSOLE] mod_commands.c:559 Timer Test: 31 sleep 20
18997
2012-08-18 17:47:25.859232 [CONSOLE] mod_commands.c:559 Timer Test: 32 sleep 20
21143
2012-08-18 17:47:25.879228 [CONSOLE] mod_commands.c:559 Timer Test: 33 sleep 20
19853
2012-08-18 17:47:25.898225 [CONSOLE] mod_commands.c:559 Timer Test: 34 sleep 20
18998
2012-08-18 17:47:25.918221 [CONSOLE] mod_commands.c:559 Timer Test: 35 sleep 20
20136
2012-08-18 17:47:25.939219 [CONSOLE] mod_commands.c:559 Timer Test: 36 sleep 20
20855
2012-08-18 17:47:25.958216 [CONSOLE] mod_commands.c:559 Timer Test: 37 sleep 20
19000
2012-08-18 17:47:25.978213 [CONSOLE] mod_commands.c:559 Timer Test: 38 sleep 20
20493
2012-08-18 17:47:25.999372 [CONSOLE] mod_commands.c:559 Timer Test: 39 sleep 20
20665
2012-08-18 17:47:26.019208 [CONSOLE] mod_commands.c:559 Timer Test: 40 sleep 20
19834
2012-08-18 17:47:26.038205 [CONSOLE] mod_commands.c:559 Timer Test: 41 sleep 20
19139
2012-08-18 17:47:26.059202 [CONSOLE] mod_commands.c:559 Timer Test: 42 sleep 20
20855
2012-08-18 17:47:26.078199 [CONSOLE] mod_commands.c:559 Timer Test: 43 sleep 20
18997
2012-08-18 17:47:26.098197 [CONSOLE] mod_commands.c:559 Timer Test: 44 sleep 20
20139
2012-08-18 17:47:26.119195 [CONSOLE] mod_commands.c:559 Timer Test: 45 sleep 20
20856
2012-08-18 17:47:26.139801 [CONSOLE] mod_commands.c:559 Timer Test: 46 sleep 20
20605
2012-08-18 17:47:26.159188 [CONSOLE] mod_commands.c:559 Timer Test: 47 sleep 20
19531
2012-08-18 17:47:26.179186 [CONSOLE] mod_commands.c:559 Timer Test: 48 sleep 20
19856
2012-08-18 17:47:26.198183 [CONSOLE] mod_commands.c:559 Timer Test: 49 sleep 20
18994
2012-08-18 17:47:26.218180 [CONSOLE] mod_commands.c:559 Timer Test: 50 sleep 20
20143

Question how can we force FS to use the sangoma timer instead of soft timer? And how can we check that FS is using Sangoma timer?
!DSPAM:5032517e32762096717492!
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




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

FreeSWITCH-dev mailing list
FreeSWITCH-dev at lists.freeswitch.org<mailto: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


!DSPAM:5032517e32762096717492!

From marketing at cluecon.com  Mon Aug 20 21:56:21 2012
From: marketing at cluecon.com (Michael Collins)
Date: Mon, 20 Aug 2012 10:56:21 -0700
Subject: [Freeswitch-dev] FreeSWITCH Weekly News and Notes
Message-ID: <CAKzWOxXKCQnH+cDshF95FruwSFTxTyc-rXq8Ww81mXXbJe-2tA@mail.gmail.com>

FreeSWITCH Weekly News and Notes is back after a brief hiatus.

In case you hadn't heard: FreeSWITCH 1.2 is out!<http://freeswitch.org/node/410>
 In fact, Anthony and Ken are working on a 1.2.2 release. Stay tuned for an
announcement. On last week's conference
call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_08_15>
 we discussed some of the git commands you may need to run in order to get
yourself moved up to the 1.2stable
branch<http://fisheye.freeswitch.org/changelog/~br=v1.2.stable/freeswitch.git>.
Join us this Wednesday<http://wiki.freeswitch.org/wiki/FS_weekly_2012_08_22>
 and we'll do a quick follow up for those who may still have questions. We
hope to have some other announcements as well.

In post-ClueCon news we'd like to let everyone know that
Vestec<http://vestec.com/>
 is finalizing the arrangements for the great ASR (automated speech
recognition) app-building contest. This is a great opportunity to get some
cash and free speech recognition licenses in return for investing some time
and effort into learning the Vestec system and building an application to
show off to the world. It's also a great way to help promote FreeSWITCH
among larger enterprises who may not realize that professional-grade ASR is
available. We will discuss this further on Wednesday's conference
call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_08_22>
.

Lastly we'd like to let everyone know that the ClueCon videos will be made
available in the coming weeks and months. Please give us some time to do a
little editing before we release them all. It will be worth the wait!

-- 
Michael S Collins
ClueCon Team
http://www.cluecon.com
877-7-4ACLUE
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From msc at freeswitch.org  Tue Aug 21 01:45:31 2012
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 20 Aug 2012 14:45:31 -0700
Subject: [Freeswitch-dev] FreeSWITCH Project Looking For Volunteers: Web
	Site Dev/Maint
Message-ID: <CAKzWOxWg2QViUwMhmj+upwTvVt0wrsxoUU4GRxw+gpkTwQVQUg@mail.gmail.com>

Hello all,

I need to refresh my list of people who have Web dev skills and are willing
to assist with things like Web site design, programming, and maintenance.
If you have any of the following skills, or know someone who does and who
is willing to donate a few hours, please email me offlist:

HTML5
CSS2/3
Javascript/JQuery
Drupal/PHP
Wordpress/PHP
Django/Python
Web design/graphics
Apache administration in LAMP environment

Please note that not every single skill is currently in demand; we are
trying to get a feel for who knows what and hopefully anticipate future
needs.

Thanks!

-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From jmesquita at freeswitch.org  Tue Aug 21 04:40:18 2012
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 20 Aug 2012 21:40:18 -0300
Subject: [Freeswitch-dev] FreeSWITCH Project Looking For Volunteers: Web
 Site Dev/Maint
In-Reply-To: <CAKzWOxWg2QViUwMhmj+upwTvVt0wrsxoUU4GRxw+gpkTwQVQUg@mail.gmail.com>
References: <CAKzWOxWg2QViUwMhmj+upwTvVt0wrsxoUU4GRxw+gpkTwQVQUg@mail.gmail.com>
Message-ID: <CABH8QX0=noTUGUtJqKAQA4LxxVJizX3roPR3Jbhs4Pu7hg4g2g@mail.gmail.com>

MC, you know you can always count on me.

Python, JQuery, Javascript and lots of willing to learn new crap. :-)

Jo?o Mesquita



On Mon, Aug 20, 2012 at 6:45 PM, Michael Collins <msc at freeswitch.org> wrote:

> Hello all,
>
> I need to refresh my list of people who have Web dev skills and are
> willing to assist with things like Web site design, programming, and
> maintenance. If you have any of the following skills, or know someone who
> does and who is willing to donate a few hours, please email me offlist:
>
> HTML5
> CSS2/3
> Javascript/JQuery
> Drupal/PHP
> Wordpress/PHP
> Django/Python
> Web design/graphics
> Apache administration in LAMP environment
>
> Please note that not every single skill is currently in demand; we are
> trying to get a feel for who knows what and hopefully anticipate future
> needs.
>
> Thanks!
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> 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
>
>
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From gabe at gundy.org  Tue Aug 21 10:32:14 2012
From: gabe at gundy.org (Gabriel Gunderson)
Date: Tue, 21 Aug 2012 00:32:14 -0600
Subject: [Freeswitch-dev] FreeSWITCH Project Looking For Volunteers: Web
 Site Dev/Maint
In-Reply-To: <CAKzWOxWg2QViUwMhmj+upwTvVt0wrsxoUU4GRxw+gpkTwQVQUg@mail.gmail.com>
References: <CAKzWOxWg2QViUwMhmj+upwTvVt0wrsxoUU4GRxw+gpkTwQVQUg@mail.gmail.com>
Message-ID: <CAMwZ-t+6xdX6wwwE7vUmvaHS9D50FN5o5iR4Qz3y+MvYzNr-Pw@mail.gmail.com>

On Mon, Aug 20, 2012 at 3:45 PM, Michael Collins <msc at freeswitch.org> wrote:
> HTML5
> CSS2/3
> Javascript/JQuery
> Django/Python
> Web design/graphics
> Apache administration in LAMP environment

The folks at Izeni are happy to help where we can (with the trimmed
list of techs).  We feel it's one of the ways we can give back to the
FreeSWITCH project and community that we benefit so much from.

We're look forward to working with the others who join in.


Best,
Gabe


From juanito1982 at gmail.com  Tue Aug 21 16:46:37 2012
From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=)
Date: Tue, 21 Aug 2012 14:46:37 +0200
Subject: [Freeswitch-dev] FreeSWITCH Project Looking For Volunteers: Web
 Site Dev/Maint
In-Reply-To: <CAKzWOxWg2QViUwMhmj+upwTvVt0wrsxoUU4GRxw+gpkTwQVQUg@mail.gmail.com>
References: <CAKzWOxWg2QViUwMhmj+upwTvVt0wrsxoUU4GRxw+gpkTwQVQUg@mail.gmail.com>
Message-ID: <CAHLYDCshuwMKjiq3v41XNzJdj24qo5BmhUqpQ511MM46a8gRgg@mail.gmail.com>

PHP, Python, Javascript/JQuery and basic HTML5/CSS23

Regards

2012/8/20 Michael Collins <msc at freeswitch.org>

> Hello all,
>
> I need to refresh my list of people who have Web dev skills and are
> willing to assist with things like Web site design, programming, and
> maintenance. If you have any of the following skills, or know someone who
> does and who is willing to donate a few hours, please email me offlist:
>
> HTML5
> CSS2/3
> Javascript/JQuery
> Drupal/PHP
> Wordpress/PHP
> Django/Python
> Web design/graphics
> Apache administration in LAMP environment
>
> Please note that not every single skill is currently in demand; we are
> trying to get a feel for who knows what and hopefully anticipate future
> needs.
>
> Thanks!
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> 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
>
>
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From msc at freeswitch.org  Wed Aug 22 01:24:44 2012
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 21 Aug 2012 14:24:44 -0700
Subject: [Freeswitch-dev] FreeSWITCH Project Looking For Volunteers: Web
	Site Dev/Maint
In-Reply-To: <CAKzWOxWg2QViUwMhmj+upwTvVt0wrsxoUU4GRxw+gpkTwQVQUg@mail.gmail.com>
References: <CAKzWOxWg2QViUwMhmj+upwTvVt0wrsxoUU4GRxw+gpkTwQVQUg@mail.gmail.com>
Message-ID: <CAKzWOxVDK0dGOFOu7W86-7UDmEz+ksm-3=yhcOcgbw_trYHy4g@mail.gmail.com>

To all those who've pinged me: many thanks! In 24 hours I've received
information from 14 different individuals - this is excellent! "Many hands
makes light work" and all that. I will definitely be in touch. In the
meantime, if anyone else has Webby skills please don't hesitate to let me
know. Also, don't forget that you can also recruit people you know who
aren't part of the FreeSWITCH community.  :)

Thanks!

-Michael

On Mon, Aug 20, 2012 at 2:45 PM, Michael Collins <msc at freeswitch.org> wrote:

> Hello all,
>
> I need to refresh my list of people who have Web dev skills and are
> willing to assist with things like Web site design, programming, and
> maintenance. If you have any of the following skills, or know someone who
> does and who is willing to donate a few hours, please email me offlist:
>
> HTML5
> CSS2/3
> Javascript/JQuery
> Drupal/PHP
> Wordpress/PHP
> Django/Python
> Web design/graphics
> Apache administration in LAMP environment
>
> Please note that not every single skill is currently in demand; we are
> trying to get a feel for who knows what and hopefully anticipate future
> needs.
>
> Thanks!
>
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From krice at freeswitch.org  Wed Aug 22 01:29:58 2012
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 21 Aug 2012 16:29:58 -0500
Subject: [Freeswitch-dev] [Freeswitch-users] FreeSWITCH Project Looking
 For Volunteers: Web Site Dev/Maint
In-Reply-To: <CAKzWOxVDK0dGOFOu7W86-7UDmEz+ksm-3=yhcOcgbw_trYHy4g@mail.gmail.com>
Message-ID: <CC596806.3C5B2%krice@freeswitch.org>

Hey also, Collins and other,

If any of you have Web Grafix/Layout skills contact me or collins... I have
a project in mind we could use a little possible help with

K


On 8/21/12 4:24 PM, "Michael Collins" <msc at freeswitch.org> wrote:

> To all those who've pinged me: many thanks! In 24 hours I've received
> information from 14 different individuals - this is excellent! "Many hands
> makes light work" and all that. I will definitely be in touch. In the
> meantime, if anyone else has Webby skills please don't hesitate to let me
> know. Also, don't forget that you can also recruit people you know who aren't
> part of the FreeSWITCH community.? :)
> 
> Thanks!
> 
> -Michael
> 
> On Mon, Aug 20, 2012 at 2:45 PM, Michael Collins <msc at freeswitch.org> wrote:
>> Hello all,
>> 
>> I need to refresh my list of people who have Web dev skills and are willing
>> to assist with things like Web site design, programming, and maintenance. If
>> you have any of the following skills, or know someone who does and who is
>> willing to donate a few hours, please email me offlist:
>> 
>> HTML5
>> CSS2/3
>> Javascript/JQuery
>> Drupal/PHP
>> Wordpress/PHP
>> Django/Python
>> Web design/graphics
>> Apache administration in LAMP environment
>> 
>> Please note that not every single skill is currently in demand; we are trying
>> to get a feel for who knows what and hopefully anticipate future needs.
>> 
>> Thanks!
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From don.dawson at voice-ring.com  Wed Aug 22 08:31:34 2012
From: don.dawson at voice-ring.com (Don Dawson)
Date: Tue, 21 Aug 2012 23:31:34 -0500
Subject: [Freeswitch-dev] SLA lights not working on some inbound calls
Message-ID: <503460A6.20709@voice-ring.com>


I have 3 Cisco SPA504G phones connected to Freeswitch using the SLA 
feature.  Everything seems to work except for inbound calls, some calls 
the lights are proper and others the lights are wrong.

I have 3 phones sharing extension 122.  The call sequence to demonstrate 
the behavior of the lights, from a fourth phone, I dial 122 and all 3 
phones ring with the x122 line key blinking.  If I answer the phone that 
is first in the list of INVITEs to x122, then the other 2 phones work 
properly with their x122 line key blinking indicating an answered call. 
However if I answer on the phone that was the 2^nd or 3^rd INVITE, then 
the line key lights for the other 2 phones go out indicating an idle 
line (it should show an active line).

Looking at the SIP trace, for the call that worked properly, there are 3 
INVITEs to the phone, 200 OK from the phone that answered (which is the 
phone that was the target of the first INVITE), CANCEL to the other 2 
phones, then the key part, the NOTIFYs are sent with 
appearance-state=active.

When I answer with one of the other 2 phones, the SIP trace shows that 
same sequence EXCEPT the NOTIFYs have appearance-state=idle.

I tried this with the latest get, version 1.2.1 and a get from about 9 
months ago.  All have the same issue.All phones register using domain name.

Here is a portion of the SLA debug showing 'idle' that when works the 
call_info_state is 'active':
2012-08-21 14:52:13.081302 [ERR] sofia_presence.c:1249 STATE SQL update 
sip_dialogs set call_info='appearance-index=1',call_info_state='idle' 
where hostname=.......................

What should I try?


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From anthony.minessale at gmail.com  Wed Aug 22 18:39:28 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 22 Aug 2012 09:39:28 -0500
Subject: [Freeswitch-dev] SLA lights not working on some inbound calls
In-Reply-To: <503460A6.20709@voice-ring.com>
References: <503460A6.20709@voice-ring.com>
Message-ID: <CAKbxfG_3N5r=V_CEbVW4inErtdM6AFEci=cUmtX4bracKdhRaA@mail.gmail.com>

open a jira with and attach a trace with the following taken from latest git

sofia global siptrace on
sofia debug sla|presence
console loglevel debug

entitle it something like cisco SLA interop


On Tue, Aug 21, 2012 at 11:31 PM, Don Dawson <don.dawson at voice-ring.com> wrote:
>
> I have 3 Cisco SPA504G phones connected to Freeswitch using the SLA feature.
> Everything seems to work except for inbound calls, some calls the lights are
> proper and others the lights are wrong.
>
>  I have 3 phones sharing extension 122.  The call sequence to demonstrate
> the behavior of the lights, from a fourth phone, I dial 122 and all 3 phones
> ring with the x122 line key blinking.  If I answer the phone that is first
> in the list of INVITEs to x122, then the other 2 phones work properly with
> their x122 line key blinking indicating an answered call.  However if I
> answer on the phone that was the 2nd or 3rd INVITE, then the line key lights
> for the other 2 phones go out indicating an idle line (it should show an
> active line).
>
>  Looking at the SIP trace, for the call that worked properly, there are 3
> INVITEs to the phone, 200 OK from the phone that answered (which is the
> phone that was the target of the first INVITE), CANCEL to the other 2
> phones, then the key part, the NOTIFYs are sent with
> appearance-state=active.
>
>  When I answer with one of the other 2 phones, the SIP trace shows that same
> sequence EXCEPT the NOTIFYs have appearance-state=idle.
>
> I tried this with the latest get, version 1.2.1 and a get from about 9
> months ago.  All have the same issue.All phones register using domain name.
>
> Here is a portion of the SLA debug showing 'idle' that when works the
> call_info_state is 'active':
> 2012-08-21 14:52:13.081302 [ERR] sofia_presence.c:1249 STATE SQL update
> sip_dialogs set call_info='appearance-index=1',call_info_state='idle' where
> hostname=.......................
>
> What should I try?
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> 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
>



-- 
Anthony Minessale II

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

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

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


From vicentini.paulo at gmail.com  Wed Aug 22 20:10:17 2012
From: vicentini.paulo at gmail.com (Paulo Vicentini)
Date: Wed, 22 Aug 2012 13:10:17 -0300
Subject: [Freeswitch-dev] FreeSwitch / M3UA
Message-ID: <CAE6hJSpPyiKgYdtsdYgUdATxfJG+mYhSv0V8qB-8McAvU9LwAg@mail.gmail.com>

Is there any plan to support M3UA in FS?

Media Gateways supports M3UA/H.248 (e.g: Audiocodes) and it would be fine
to interoperate directly with them with FS:

FS (SGC)<---SIGTRAN (M3UA)/Megaco ---> AudioCodes SGW (MTP1,2)<----link set
ss7

FS would need a Sigtran/M3UA module to communicate with signaling gateway
(AudioCodes M8k) and a Megaco(H.248) module to control audio MGW
(Notice that MTP1,2 isn't handled in FS  - M3UA(ISUP) )

Any comments on that?

Thanks
Paulo
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From msc at freeswitch.org  Wed Aug 22 20:26:03 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 22 Aug 2012 09:26:03 -0700
Subject: [Freeswitch-dev] FreeSWITCH Community Conference Call Today
 (Important Information - Please Read)
Message-ID: <CAKzWOxX0_c41zoW7y3JHqP33tSV0xe4Pmcp_7ioddTdVMw_qvg@mail.gmail.com>

Hello all!

The conference will start in less than an hour. Here's our agenda:

http://wiki.freeswitch.org/wiki/FS_weekly_2012_08_22

We've made a change to the system. Everyone will come in muted. When you
press 0 it will send a message into the #FreeSWITCH IRC channel - "user has
a question". One of the conference moderators will unmute you so that you
can ask your question. We are hoping this will make the conference more
enjoyable for everyone.

Talk to you soon!

-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From mike at jerris.com  Wed Aug 22 20:46:13 2012
From: mike at jerris.com (Michael Jerris)
Date: Wed, 22 Aug 2012 12:46:13 -0400
Subject: [Freeswitch-dev] Creating a session without incoming or
	outgoing call
In-Reply-To: <CABuGx5MUsD0fbz42e3V1_AzEcOu8mSEXj08+Uk9zcCN3R9BUog@mail.gmail.com>
References: <CABuGx5MUsD0fbz42e3V1_AzEcOu8mSEXj08+Uk9zcCN3R9BUog@mail.gmail.com>
Message-ID: <B19DC1F0-A1E4-4582-89FD-05B250D60DDE@jerris.com>

It's all software, anything is possible.  I guess the question is why do you want to do these things.  You could probably do all this from the api interface or app.

Mike

On Aug 7, 2012, at 3:09 AM, saurabh nirkhey <snirkhey at gmail.com> wrote:

> I am using mod_skel and writing my own module. My application needs me
> to be able to create a session without an incoming call. My questions
> are:
> 
> 1. Is it possible to do this?
> 2. Do I need to inherit CoreSession or does an easier alternative exist?
> 3. Can I invoke my application from eventsocket without originating a
> call at all?
> 
> Thanks for any inputs.



From vutamhoan at gmail.com  Thu Aug 23 06:02:47 2012
From: vutamhoan at gmail.com (Vu Quang Hoa)
Date: Thu, 23 Aug 2012 09:02:47 +0700
Subject: [Freeswitch-dev] FreeSwitch / M3UA
In-Reply-To: <CAE6hJSpPyiKgYdtsdYgUdATxfJG+mYhSv0V8qB-8McAvU9LwAg@mail.gmail.com>
References: <CAE6hJSpPyiKgYdtsdYgUdATxfJG+mYhSv0V8qB-8McAvU9LwAg@mail.gmail.com>
Message-ID: <CAJnMD2e=w-Q+bG4YiU_KrKXW-u2dXZUxxH61Sd0ri65gW5YitQ@mail.gmail.com>

I suggest Mobicents instead of FS in this scenario. Mobicents is dedicated
for telecom signaling (M3UA and Megaco is ready, even more protocols)

On Wed, Aug 22, 2012 at 11:10 PM, Paulo Vicentini <vicentini.paulo at gmail.com
> wrote:

> Is there any plan to support M3UA in FS?
>
> Media Gateways supports M3UA/H.248 (e.g: Audiocodes) and it would be fine
> to interoperate directly with them with FS:
>
> FS (SGC)<---SIGTRAN (M3UA)/Megaco ---> AudioCodes SGW (MTP1,2)<----link
> set ss7
>
> FS would need a Sigtran/M3UA module to communicate with signaling gateway
> (AudioCodes M8k) and a Megaco(H.248) module to control audio MGW
> (Notice that MTP1,2 isn't handled in FS  - M3UA(ISUP) )
>
> Any comments on that?
>
> Thanks
> Paulo
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> 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
>
>
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From SPapineni at enghouse.com  Thu Aug 23 18:56:01 2012
From: SPapineni at enghouse.com (Papineni, Suneel)
Date: Thu, 23 Aug 2012 14:56:01 +0000
Subject: [Freeswitch-dev] Channel_Bridge event
Message-ID: <9438D04074E0DE45A49CD760998212725BEE0D78@CORP-MAIL-001.edge.local>

Hi,

I am expecting a "Channel_Bridge" event when I issued "uuid_bridge" command between two call legs (one call is active and answered and other call is in ringing state), but I am not getting this event. I tried using fs_cli.exe to view the events. Only when other party answered, then I could see Channel_Bridge event.

Can someone help me in understanding why I am not getting this event when Bridge command is issued?

As per uuid_bridge command wiki, call should get bridged if at least one party is answered and active.

Thanks& Regards
Suneel

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From peter.olsson at visionutveckling.se  Thu Aug 23 19:08:08 2012
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 23 Aug 2012 15:08:08 +0000
Subject: [Freeswitch-dev] Channel_Bridge event
Message-ID: <1FFF97C269757C458224B7C895F35F1514CE3D@cantor.std.visionutv.se>

I believe the event only arrives when they are actually talking to each other, and that is when both parties are answered. However, it's totally valid to do uuid_bridge if at least one call is answered.

/Peter

Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] F?r Papineni, Suneel
Skickat: den 23 augusti 2012 16:56
Till: freeswitch-dev at lists.freeswitch.org
?mne: [Freeswitch-dev] Channel_Bridge event

Hi,

I am expecting a "Channel_Bridge" event when I issued "uuid_bridge" command between two call legs (one call is active and answered and other call is in ringing state), but I am not getting this event. I tried using fs_cli.exe to view the events. Only when other party answered, then I could see Channel_Bridge event.

Can someone help me in understanding why I am not getting this event when Bridge command is issued?

As per uuid_bridge command wiki, call should get bridged if at least one party is answered and active.

Thanks& Regards
Suneel

!DSPAM:503641c532762417717798!
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From vicentini.paulo at gmail.com  Thu Aug 23 19:29:20 2012
From: vicentini.paulo at gmail.com (Paulo Vicentini)
Date: Thu, 23 Aug 2012 12:29:20 -0300
Subject: [Freeswitch-dev] FreeSwitch / M3UA
In-Reply-To: <CAJnMD2e=w-Q+bG4YiU_KrKXW-u2dXZUxxH61Sd0ri65gW5YitQ@mail.gmail.com>
References: <CAE6hJSpPyiKgYdtsdYgUdATxfJG+mYhSv0V8qB-8McAvU9LwAg@mail.gmail.com>
	<CAJnMD2e=w-Q+bG4YiU_KrKXW-u2dXZUxxH61Sd0ri65gW5YitQ@mail.gmail.com>
Message-ID: <CAE6hJSqeGyx3a3hRi5EQK1QO3-r68srSU49JdMZJe57wERa+tw@mail.gmail.com>

Have you already tried Mobicents with AudioCodes (m3ua/megaco)?

PS:
It would be great to have mod_megaco and mod_sigtran_client on FS
(Softswitch)

Paulo

On Wed, Aug 22, 2012 at 11:02 PM, Vu Quang Hoa <vutamhoan at gmail.com> wrote:

> I suggest Mobicents instead of FS in this scenario. Mobicents is dedicated
> for telecom signaling (M3UA and Megaco is ready, even more protocols)
>
> On Wed, Aug 22, 2012 at 11:10 PM, Paulo Vicentini <
> vicentini.paulo at gmail.com> wrote:
>
>> Is there any plan to support M3UA in FS?
>>
>> Media Gateways supports M3UA/H.248 (e.g: Audiocodes) and it would be fine
>> to interoperate directly with them with FS:
>>
>> FS (SGC)<---SIGTRAN (M3UA)/Megaco ---> AudioCodes SGW (MTP1,2)<----link
>> set ss7
>>
>> FS would need a Sigtran/M3UA module to communicate with signaling gateway
>> (AudioCodes M8k) and a Megaco(H.248) module to control audio MGW
>> (Notice that MTP1,2 isn't handled in FS  - M3UA(ISUP) )
>>
>> Any comments on that?
>>
>> Thanks
>> Paulo
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> 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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> 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
>
>
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From ncorbic at sangoma.com  Thu Aug 23 18:18:49 2012
From: ncorbic at sangoma.com (Nenad Corbic)
Date: Thu, 23 Aug 2012 10:18:49 -0400
Subject: [Freeswitch-dev] FreeSwitch / M3UA
In-Reply-To: <CAJnMD2e=w-Q+bG4YiU_KrKXW-u2dXZUxxH61Sd0ri65gW5YitQ@mail.gmail.com>
References: <CAE6hJSpPyiKgYdtsdYgUdATxfJG+mYhSv0V8qB-8McAvU9LwAg@mail.gmail.com>
	<CAJnMD2e=w-Q+bG4YiU_KrKXW-u2dXZUxxH61Sd0ri65gW5YitQ@mail.gmail.com>
Message-ID: <F9F12310C0FDC54096D91B463439C3C90481DBCDA6@IAD2MBX19.mex02.mlsrvr.com>

Sangoma has M3UA in FS :)
It will be part of our new NSG product.
If you are interested plz contact me at ncorbic at sangoma.com

Nenad



Nenad Corbic, B.Eng
VP Software Engineering
ncorbic at sangoma.com<mailto:ncorbic at sangoma.com>
Sangoma Technologies
100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada


t.   +1 800 388 2475 (N. America)
t.   +1 905 474 1990 x113
f.   +1 905 474 9223


[cid:image001.jpg at 01CD8118.AFB64A50]<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email+signatures>

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                VegaStream is now part of Sangoma!<http://sangoma.com/about_us/newsroom/news_releases/news/1488795/Sangoma_Acquires_All_Key_Assets_of_VegaStream.html>
                Ask us about both Gateway Appliances<http://sangoma.com/products/media_gateways/vega.html> and Internal Gateways<http://sangoma.com/products/software_products/netborder_suite/netborder_express.html>


From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Vu Quang Hoa
Sent: Wednesday, August 22, 2012 10:03 PM
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] FreeSwitch / M3UA

I suggest Mobicents instead of FS in this scenario. Mobicents is dedicated for telecom signaling (M3UA and Megaco is ready, even more protocols)
On Wed, Aug 22, 2012 at 11:10 PM, Paulo Vicentini <vicentini.paulo at gmail.com<mailto:vicentini.paulo at gmail.com>> wrote:
Is there any plan to support M3UA in FS?

Media Gateways supports M3UA/H.248 (e.g: Audiocodes) and it would be fine to interoperate directly with them with FS:

FS (SGC)<---SIGTRAN (M3UA)/Megaco ---> AudioCodes SGW (MTP1,2)<----link set ss7

FS would need a Sigtran/M3UA module to communicate with signaling gateway (AudioCodes M8k) and a Megaco(H.248) module to control audio MGW
(Notice that MTP1,2 isn't handled in FS  - M3UA(ISUP) )

Any comments on that?

Thanks
Paulo

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




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

FreeSWITCH-dev mailing list
FreeSWITCH-dev at lists.freeswitch.org<mailto:FreeSWITCH-dev at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
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From alhakeem at ipextelecom.net  Thu Aug 23 19:58:47 2012
From: alhakeem at ipextelecom.net (Abdul Hakeem)
Date: Thu, 23 Aug 2012 16:58:47 +0100
Subject: [Freeswitch-dev] FreeSwitch / M3UA
In-Reply-To: <F9F12310C0FDC54096D91B463439C3C90481DBCDA6@IAD2MBX19.mex02.mlsrvr.com>
References: <CAE6hJSpPyiKgYdtsdYgUdATxfJG+mYhSv0V8qB-8McAvU9LwAg@mail.gmail.com>	<CAJnMD2e=w-Q+bG4YiU_KrKXW-u2dXZUxxH61Sd0ri65gW5YitQ@mail.gmail.com>
	<F9F12310C0FDC54096D91B463439C3C90481DBCDA6@IAD2MBX19.mex02.mlsrvr.com>
Message-ID: <!&!AAAAAAAAAAAYAAAAAAAAAKU8ZfJ4xhNEj1obo+caIU8ihAAAEAAAAEitUdP8YBZJi0tE2ckaJQwBAAAAAA==@ipextelecom.net>

Nenad,
 
Would you be kind enough to forward details of the NSG product ?
 
Regards,
Abdul Hakeem
 
 
From: freeswitch-dev-bounces at lists.freeswitch.org
[mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Nenad Corbic
Sent: Thursday, August 23, 2012 3:19 PM
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] FreeSwitch / M3UA
 
Sangoma has M3UA in FS J
It will be part of our new NSG product.
If you are interested plz contact me at ncorbic at sangoma.com
 
Nenad
 
 
 

Nenad Corbic, B.Eng
VP Software Engineering
ncorbic at sangoma.com
Sangoma Technologies
100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada 

t.   +1 800 388 2475 (N. America)
t.   +1 905 474 1990 x113
f.   +1 905 474 9223
 
 
<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campai
gn=email+signatures> Description: Description:
cid:image001.jpg at 01CC81EE.A6D44330

 
<http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campaign=
email%2Bsignatures> Products |
<http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_campaign
=email%2Bsignatures> Solutions |
<http://sangoma.com/about_us/events?utm_source=signature&utm_medium=email&utm_ca
mpaign=email%2Bsignatures> Events |
<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campai
gn=email%2Bsignatures> Contact |
<http://wiki.sangoma.com/?utm_source=signature&utm_medium=email&utm_campaign=ema
il%2Bsignatures> Wiki |
<http://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=signatu
re&utm_medium=email&utm_campaign=email%2Bsignatures> Facebook |
<http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_campai
gn=email%2Bsignatures> Twitter |
<http://www.youtube.com/sangomatechnologies?utm_source=signature&utm_medium=emai
l&utm_campaign=email%2Bsignatures> YouTube
                VegaStream is now part of Sangoma!
<http://sangoma.com/about_us/newsroom/news_releases/news/1488795/Sangoma_Acquire
s_All_Key_Assets_of_VegaStream.html> 
                Ask us about both Gateway Appliances
<http://sangoma.com/products/media_gateways/vega.html>  and Internal Gateways
<http://sangoma.com/products/software_products/netborder_suite/netborder_express
.html> 
 
 
From: freeswitch-dev-bounces at lists.freeswitch.org
[mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Vu Quang Hoa
Sent: Wednesday, August 22, 2012 10:03 PM
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] FreeSwitch / M3UA
 
I suggest Mobicents instead of FS in this scenario. Mobicents is dedicated for
telecom signaling (M3UA and Megaco is ready, even more protocols)
On Wed, Aug 22, 2012 at 11:10 PM, Paulo Vicentini <vicentini.paulo at gmail.com>
wrote:
Is there any plan to support M3UA in FS? 
 
Media Gateways supports M3UA/H.248 (e.g: Audiocodes) and it would be fine to
interoperate directly with them with FS:
 
FS (SGC)<---SIGTRAN (M3UA)/Megaco ---> AudioCodes SGW (MTP1,2)<----link set ss7
 
FS would need a Sigtran/M3UA module to communicate with signaling gateway
(AudioCodes M8k) and a Megaco(H.248) module to control audio MGW  
(Notice that MTP1,2 isn't handled in FS  - M3UA(ISUP) )
 
Any comments on that?
 
Thanks
Paulo

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




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

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
 
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From vutamhoan at gmail.com  Fri Aug 24 03:17:18 2012
From: vutamhoan at gmail.com (Vu Quang Hoa)
Date: Fri, 24 Aug 2012 06:17:18 +0700
Subject: [Freeswitch-dev] FreeSwitch / M3UA
In-Reply-To: <CAE6hJSqeGyx3a3hRi5EQK1QO3-r68srSU49JdMZJe57wERa+tw@mail.gmail.com>
References: <CAE6hJSpPyiKgYdtsdYgUdATxfJG+mYhSv0V8qB-8McAvU9LwAg@mail.gmail.com>
	<CAJnMD2e=w-Q+bG4YiU_KrKXW-u2dXZUxxH61Sd0ri65gW5YitQ@mail.gmail.com>
	<CAE6hJSqeGyx3a3hRi5EQK1QO3-r68srSU49JdMZJe57wERa+tw@mail.gmail.com>
Message-ID: <CAJnMD2cumwF==dcMGDDZsbUJsOaDYyjDADOL+eErv_wUDnP+xw@mail.gmail.com>

Yes, I wrote a MGC and run with Median 2000. FS was media server

Right tool for the right job. mod_sigtran_client would cost alot of effort
and it's not easy to achieve

On Thu, Aug 23, 2012 at 10:29 PM, Paulo Vicentini <vicentini.paulo at gmail.com
> wrote:

> Have you already tried Mobicents with AudioCodes (m3ua/megaco)?
>
> PS:
> It would be great to have mod_megaco and mod_sigtran_client on FS
> (Softswitch)
>
> Paulo
>
> On Wed, Aug 22, 2012 at 11:02 PM, Vu Quang Hoa <vutamhoan at gmail.com>wrote:
>
>> I suggest Mobicents instead of FS in this scenario. Mobicents is
>> dedicated for telecom signaling (M3UA and Megaco is ready, even more
>> protocols)
>>
>> On Wed, Aug 22, 2012 at 11:10 PM, Paulo Vicentini <
>> vicentini.paulo at gmail.com> wrote:
>>
>>> Is there any plan to support M3UA in FS?
>>>
>>> Media Gateways supports M3UA/H.248 (e.g: Audiocodes) and it would be
>>> fine to interoperate directly with them with FS:
>>>
>>> FS (SGC)<---SIGTRAN (M3UA)/Megaco ---> AudioCodes SGW (MTP1,2)<----link
>>> set ss7
>>>
>>> FS would need a Sigtran/M3UA module to communicate with signaling
>>> gateway (AudioCodes M8k) and a Megaco(H.248) module to control audio MGW
>>> (Notice that MTP1,2 isn't handled in FS  - M3UA(ISUP) )
>>>
>>> Any comments on that?
>>>
>>> Thanks
>>> Paulo
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> 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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> 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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> 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
>
>
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From sdame at 207me.com  Fri Aug 24 15:15:02 2012
From: sdame at 207me.com (Stephen Dame)
Date: Fri, 24 Aug 2012 07:15:02 -0400
Subject: [Freeswitch-dev] Scaled or distrbuted mod_conference?
Message-ID: <00ba01cd81e9$b5523880$1ff6a980$@207me.com>

 

I know a few commercial companies have scaled freeswitch to handle 250-500
callers in a single conference across boxes with direct audio cable
connections in data center.

 

I have an existing application that uses speex16(flash voip) and I can get
50 callers in a single conference before cpu gets to 70% on a c1.medium
instance.   I know I can throw more hardware at the problem,  but interested
in bridging the same conference number between multiple freeswitch instances
and presenting single ESL notifications back to the existing application so
all users events are seen.

 

This is a distance learning app, and not business audio conference, so some
latency is tolerable.   I would locate the multiple freeswitch servers in
same zone and possibly use the new high I/O EBS instances running in SSD.

 

Any thoughts on this?

 

Regards,

Stephen

 

 

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From shaheryarkh at googlemail.com  Fri Aug 24 19:59:18 2012
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Fri, 24 Aug 2012 17:59:18 +0200
Subject: [Freeswitch-dev] Scaled or distrbuted mod_conference?
In-Reply-To: <00ba01cd81e9$b5523880$1ff6a980$@207me.com>
References: <00ba01cd81e9$b5523880$1ff6a980$@207me.com>
Message-ID: <CAFZQphzxstRwgo2G=f4FMTsh46NqFu5ja7sP2XAh06Ci=+4T6w@mail.gmail.com>

hummm, interesting, i was just wondering if it is possible to make a tree
shape conferencing bridge which would work something like this.

1. We will have many freeswitch boxes, each will running a single
conference of its own up to say 50 users. These freeswitch box may be
geographically distributed as needed, (though it may cause significant
latency).

2. Then there is a freeswitch box running on top, all freeswitch boxes dial
this top level freeswitch box as normal call. On top level freeswitch these
calls will be answered and joined in a single so called top level
conference. The second leg of this call will be joined to conference
already running in lower level freeswitch boxes.

3. The main speaker / teacher of this virtual class room will just dial to
top level freeswitch and join in the top level conference..

4. For listeners / students of this virtual class to dial in and join, we
can setup an OpenSIPs / Kamailio boxes which authenticates users first then
forwards call to lower level freeswitch boxes.

5. All student calls will be mute when they join the conference. The
teacher / speaker will have a web management interface which will query
each lower freeswitch box and generate a participants list. Teacher will be
able to unmute any student to listen to their questions etc. and then mute
them again to answer the question etc.

What do you guys think? Please suggest.

Thank you.


On Fri, Aug 24, 2012 at 1:15 PM, Stephen Dame <sdame at 207me.com> wrote:

> ** **
>
> *I know a few commercial companies have scaled freeswitch to handle
> 250-500 callers in a single conference across boxes with direct audio cable
> connections in data center.*
>
> * *
>
> * I have an existing application that uses speex16(flash voip) and I can
> get 50 callers in a single conference before cpu gets to 70% on a c1.medium
> instance.   I know I can throw more hardware at the problem,  but
> interested in bridging the same conference number between multiple
> freeswitch instances and presenting single ESL notifications back to the
> existing application so all users events are seen.*
>
> * *
>
> *This is a distance learning app, and not business audio conference, so
> some latency is tolerable.   I would locate the multiple freeswitch servers
> in same zone and possibly use the new high I/O EBS instances running in SSD.
> *
>
> * *
>
> *Any thoughts on this?*
>
> * *
>
> *Regards,*
>
> *Stephen*
>
> * *
>
> * *
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-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
>
>


-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From william.king at quentustech.com  Fri Aug 24 20:24:40 2012
From: william.king at quentustech.com (William King)
Date: Fri, 24 Aug 2012 09:24:40 -0700
Subject: [Freeswitch-dev] Scaled or distrbuted mod_conference?
In-Reply-To: <00ba01cd81e9$b5523880$1ff6a980$@207me.com>
References: <00ba01cd81e9$b5523880$1ff6a980$@207me.com>
Message-ID: <5037AAC8.4080107@quentustech.com>

You might be able to setup a single top node conference, then bridge in
lower node conferences. In this situation you could have 6 lower node
boxes each with 50 callers into it, and each of the 6 lower node
conferences called into the single top node conference(along with your
main audio producing stream(the professor, etc). This would be a bit
more complicated to control for muting, etc.

William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337
Office: (206) 388-4772
Cell:   (253) 686-5518
william.king at quentustech.com


On 08/24/2012 04:15 AM, Stephen Dame wrote:
>
>  
>
> *I know a few commercial companies have scaled freeswitch to handle
> 250-500 callers in a single conference across boxes with direct audio
> cable connections in data center.*
>
> * *
>
> *I have an existing application that uses speex16(flash voip) and I
> can get 50 callers in a single conference before cpu gets to 70% on a
> c1.medium instance.   I know I can throw more hardware at the
> problem,  but interested in bridging the same conference number
> between multiple freeswitch instances and presenting single ESL
> notifications back to the existing application so all users events are
> seen.*
>
> * *
>
> *This is a distance learning app, and not business audio conference,
> so some latency is tolerable.   I would locate the multiple freeswitch
> servers in same zone and possibly use the new high I/O EBS instances
> running in SSD.*
>
> * *
>
> *Any thoughts on this?*
>
> * *
>
> *Regards,*
>
> *Stephen*
>
> * *
>
> * *
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-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
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From czimm at goodsstores.com  Fri Aug 24 21:30:36 2012
From: czimm at goodsstores.com (Craig N. Zimmerman)
Date: Fri, 24 Aug 2012 13:30:36 -0400
Subject: [Freeswitch-dev] FreeSWITCH-dev Digest, Vol 74, Issue 12
In-Reply-To: <mailman.17586.1345823990.13443.freeswitch-dev@lists.freeswitch.org>
References: <mailman.17586.1345823990.13443.freeswitch-dev@lists.freeswitch.org>
Message-ID: <DF10BBD8D04F8C479E9F898A25CA3A973F973B287C@dc-mail>

Hi Steve

We have successfully done this with 900+ callers in a little different setup. Here is how we do it: The call comes into the first fs box then at about 300-350 callers. I have a script that sends a sip call to another box so we have an audio link then we forward all additional incoming calls to that box till with bypass media till it is maxed out then to the next box and then the next. You should be able to get up to 400 callers on one box depending on your hardware, bandwidth and codec. Also it helps to rent dedicated boxed to deal with the timing issues.

Also we don't do the web interface but you can do that with a mysql db to keep track of which box the call is on.

The only reason we have to this is because we can't get enough bandwidth into a single box. The other thing that I would like to do some time with this setup is hook my scripts up to rackspace or amazon's api and spin up more servers after I run out of physical boxes during peak load times.

Thanks
Craig
I hope this helps.

-----Original Message-----
From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of freeswitch-dev-request at lists.freeswitch.org
Sent: Friday, August 24, 2012 12:00 PM
To: freeswitch-dev at lists.freeswitch.org
Subject: FreeSWITCH-dev Digest, Vol 74, Issue 12

hummm, interesting, i was just wondering if it is possible to make a tree shape conferencing bridge which would work something like this.

1. We will have many freeswitch boxes, each will running a single conference of its own up to say 50 users. These freeswitch box may be geographically distributed as needed, (though it may cause significant latency).

2. Then there is a freeswitch box running on top, all freeswitch boxes dial this top level freeswitch box as normal call. On top level freeswitch these calls will be answered and joined in a single so called top level conference. The second leg of this call will be joined to conference already running in lower level freeswitch boxes.

3. The main speaker / teacher of this virtual class room will just dial to top level freeswitch and join in the top level conference..

4. For listeners / students of this virtual class to dial in and join, we can setup an OpenSIPs / Kamailio boxes which authenticates users first then forwards call to lower level freeswitch boxes.

5. All student calls will be mute when they join the conference. The teacher / speaker will have a web management interface which will query each lower freeswitch box and generate a participants list. Teacher will be able to unmute any student to listen to their questions etc. and then mute them again to answer the question etc.

What do you guys think? Please suggest.

Thank you.


On Fri, Aug 24, 2012 at 1:15 PM, Stephen Dame <sdame at 207me.com> wrote:
 
I know a few commercial companies have scaled freeswitch to handle 250-500 callers in a single conference across boxes with direct audio cable connections in data center.
 
I have an existing application that uses speex16(flash voip) and I can get 50 callers in a single conference before cpu gets to 70% on a c1.medium instance.   I know I can throw more hardware at the problem,  but interested in bridging the same conference number between multiple freeswitch instances and presenting single ESL notifications back to the existing application so all users events are seen.
 
This is a distance learning app, and not business audio conference, so some latency is tolerable.   I would locate the multiple freeswitch servers in same zone and possibly use the new high I/O EBS instances running in SSD.
 
Any thoughts on this?
 
Regards,
Stephen
 
 

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




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

FreeSWITCH-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




-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com


From saju.pillai at gmail.com  Mon Aug 27 08:24:32 2012
From: saju.pillai at gmail.com (Saju Pillai)
Date: Mon, 27 Aug 2012 09:54:32 +0530
Subject: [Freeswitch-dev] libfreeswitch as embedded IVR
Message-ID: <CAM7Ljg-DabT6cs=UVvuXoXDMF_JBXK6BcjCiTqgJihq2WsfHOg@mail.gmail.com>

Hello,

 I have to implement an IVR embedded in an existing app. My
understanding is that I can declare global handlers for channel states
to *intercept* calls and use the switch_ivr_* to do my processing. I
have to either use tts or play recorded prompts to the caller and
receive dtmf input from the caller. There can be multiple to & fro
interactions with the caller in 1 call, this is determined by other
data from the rest of the app.

1) Which channel state(s) should I hook into?
2) Will I have to setup a dialplan to make this work ?
3) Is there a sample/test or psuedo code that I can look at ?

regards
srp


From gvvsubhashkumar at gmail.com  Mon Aug 27 19:19:44 2012
From: gvvsubhashkumar at gmail.com (Subhash)
Date: Mon, 27 Aug 2012 20:49:44 +0530
Subject: [Freeswitch-dev] How to write an Application to control freeSWITCH
Message-ID: <CAFvodxU5RQx4VGQ4uUyoBw22fstppmwoNUWAE5k0x=+HXdj2NA@mail.gmail.com>

Hi All,

       I am new to freeSWITCH platform.

      I want to write an application using c++ to control freeSWITCH so
please guide me.

      Thanks in advance.

Thanks,
Subhash.
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From aroumie at yahoo.com  Tue Aug 28 01:29:43 2012
From: aroumie at yahoo.com (Ali R.)
Date: Mon, 27 Aug 2012 14:29:43 -0700 (PDT)
Subject: [Freeswitch-dev] How to write an Application to control
	freeSWITCH
In-Reply-To: <CAFvodxU5RQx4VGQ4uUyoBw22fstppmwoNUWAE5k0x=+HXdj2NA@mail.gmail.com>
References: <CAFvodxU5RQx4VGQ4uUyoBw22fstppmwoNUWAE5k0x=+HXdj2NA@mail.gmail.com>
Message-ID: <1346102983.67867.YahooMailNeo@web120306.mail.ne1.yahoo.com>

I can only speak for my experience with
FS. As you mention FreeSwitch is a platform that means not a basic app. I recommend
starting by reading the 2 books about FreeSwitch if you don't like to go
through the Wiki jungle. Once you understand the main modules and the flow of
events you can move on?to the next phase where you can either write a module in
C++ to control it from within the same process or use the powerful event socket
in inbound mode for total control from anywhere else...
 

________________________________
 From: Subhash <gvvsubhashkumar at gmail.com>
To: Subhash kumar <gvvsubhashkumar at gmail.com> 
Sent: Monday, August 27, 2012 8:19 AM
Subject: [Freeswitch-dev] How to write an Application to control freeSWITCH
  


Hi All,

? ? ? ?I am new to freeSWITCH platform.

? ? ? I want to write an application using c++ to control freeSWITCH so please guide me. 
? ? ?
? ? ? Thanks in advance.

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


/

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

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/
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From aroumie at yahoo.com  Tue Aug 28 08:26:56 2012
From: aroumie at yahoo.com (Ali R.)
Date: Mon, 27 Aug 2012 21:26:56 -0700 (PDT)
Subject: [Freeswitch-dev] How to write an Application to control
	freeSWITCH
In-Reply-To: <CAFvodxWcs0scztq4sfxagtyaEHhhFmd_PmMVa3jeRVZHNz3WPA@mail.gmail.com>
References: <CAFvodxU5RQx4VGQ4uUyoBw22fstppmwoNUWAE5k0x=+HXdj2NA@mail.gmail.com>
	<1346102983.67867.YahooMailNeo@web120306.mail.ne1.yahoo.com>
	<CAFvodxWcs0scztq4sfxagtyaEHhhFmd_PmMVa3jeRVZHNz3WPA@mail.gmail.com>
Message-ID: <1346128016.63308.YahooMailNeo@web120301.mail.ne1.yahoo.com>

http://www.amazon.com/FreeSWITCH-1-0-6-Anthony-Minessale/dp/1847199968/ref=sr_1_1?ie=UTF8&qid=1346127872&sr=8-1&keywords=freeswitch
http://www.amazon.com/FreeSWITCH-Cookbook-Anthony-Minessale/dp/1849515409/ref=pd_sim_b_2
?
 

________________________________
 From: Subhash <gvvsubhashkumar at gmail.com>
To: aroumie at yahoo.com 
Sent: Monday, August 27, 2012 8:16 PM
Subject: Re: [Freeswitch-dev] How to write an Application to control freeSWITCH
  

Hi,

Thanks for the reply.

Can you please name the books and where can i get them.


Thanks,
Subhash.



On Tue, Aug 28, 2012 at 2:59 AM, Ali R. <aroumie at yahoo.com> wrote:

I can only speak for my experience with
FS. As you mention FreeSwitch is a platform that means not a basic app. I recommend
starting by reading the 2 books about FreeSwitch if you don't like to go
through the Wiki jungle. Once you understand the main modules and the flow of
events you can move on?to the next phase where you can either write a module in
C++ to control it from within the same process or use the powerful event socket
in inbound mode for total control from anywhere else...
>
> 
> From: Subhash <gvvsubhashkumar at gmail.com>
>To: Subhash kumar <gvvsubhashkumar at gmail.com> 
>Sent: Monday, August 27, 2012 8:19 AM
>Subject: [Freeswitch-dev] How to write an Application to control freeSWITCH
>  
>
>
>
>Hi All,
>
>
>? ? ? ?I am new to freeSWITCH platform.
>
>
>? ? ? I want to write an application using c++ to control freeSWITCH so please guide me. 
>? ? ?
>? ? ? Thanks in advance.
>
>Thanks,
>Subhash.
> 
>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com/
>
>
>
>/
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org/
>http://wiki.freeswitch.org/
>http://www.cluecon.com/
>
>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/
>
>
>  
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.com
>
>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/
>
>
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From marketing at cluecon.com  Wed Aug 29 01:07:35 2012
From: marketing at cluecon.com (Michael Collins)
Date: Tue, 28 Aug 2012 14:07:35 -0700
Subject: [Freeswitch-dev] FreeSWITCH Weekly News and Notes
Message-ID: <CAKzWOxVtqAVedgWH=H0XEpW+0bZ+0KiaNKhUPOzi-ZQfyjETmw@mail.gmail.com>

Hello all!

Apologies for the delay on this week's news and notes. Yesterday was quite
busy, but things are going well. I am happy to report that we have
submitted the first chapter of the new FreeSWITCH book to the publisher!
We've also conferred with a few members of the community and convinced
Packt<http://www.packtpub.com>to let us add some bonus content! Stay
tuned for more previews and teases.
Right now it's still early in the game so I don't want to reveal too much.

On last week's conference
call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_08_22>we discussed
a number of things. First, we did a follow up to Ken's
previous discussion
<http://wiki.freeswitch.org/wiki/FS_weekly_2012_08_15>about the stable
1.2 branch and using git. Second, we talked a bit about
Vestec <http://www.vestec.com> and the great ASR application contest. (More
information is forthcoming!) Lastly, we had Mitch Capper discussing the
latest version of the FSClient
<http://wiki.freeswitch.org/wiki/FSClient>Windows softphone. If you
haven't tried it out I highly recommend it. It's
now stable and feature-enabled to the point that I've discontinued using
X-Lite or Jitsi.

Tomorrow we hope to have a discussion about
TLS<http://wiki.freeswitch.org/wiki/Tls>.
We have several community members who are experienced with key and
certificate management and we will be calling upon them to share their
experience with the rest of us. After that we will have an open discussion.

Cheers!

-- 
Michael S Collins
ClueCon Team
http://www.cluecon.com
877-7-4ACLUE
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From msc at freeswitch.org  Wed Aug 29 20:09:45 2012
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 29 Aug 2012 09:09:45 -0700
Subject: [Freeswitch-dev] FreeSWITCH Conference Call Today
Message-ID: <CAKzWOxXyEZWCvDfW=MEKvXcFG1hKJQdb=Xw8WOrQ_+i3HmgWHg@mail.gmail.com>

Hello all,

We are having a nice discussion today:

http://wiki.freeswitch.org/wiki/FS_weekly_2012_08_22

Mitch Capper and others who have experience will be talking to us about
TLS, certs, CA's and things like that. Hopefully we'll all be better
prepared to enable TLS transport on our devices and help those who come to
the IRC and ML with TLS questions.

Talk to you soon!

-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From fdelawarde at wirelessmundi.com  Wed Aug 29 20:43:30 2012
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Wed, 29 Aug 2012 18:43:30 +0200
Subject: [Freeswitch-dev] [Freeswitch-users] Doubts about T38 passthru
In-Reply-To: <1346229988.23367.51.camel@luna.madrid.commsmundi.com>
References: <1346229988.23367.51.camel@luna.madrid.commsmundi.com>
Message-ID: <1346258610.1241.12.camel@luna.madrid.commsmundi.com>

Hi,

Sent to the wrong ML, this probably is more a dev question:

On Wed, 2012-08-29 at 10:46 +0200, Fran?ois Delawarde wrote: 
> Hi,
> 
> Whenever my provider puts me on hold, it reINVITES with an SDP
> containing both audio (sendonly) and image media like so:
> 
> ...
> m=audio 5320 RTP/AVP 18 3 8 0 96
> c=IN IP4 <ip_address>
> a=rtpmap:18 G729/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=sendonly
> a=rtpmap:96 telephone-event/8000
> m=image 5322 udptl t38
> c=IN IP4 <ip_address>
> a=T38FaxVersion:0
> a=T38MaxBitRate:14400
> a=T38FaxUdpEC:t38UDPRedundancy
> a=T38FaxRateManagement:transferredTCF
> 
> 
> Not sure if it's a non-standard SDP, but that's what I get. Now when
> t38_passthru=true, it only forwards the "image" part to the other leg.
> When t38_passthru=false, it only forwards the "audio" part.
> 
> Is there a way to make t38_passthru forward the audio part as well? Or
> should I just change to a better provider?

Thanks,
Fran?ois.



From anthony.minessale at gmail.com  Wed Aug 29 20:48:16 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 29 Aug 2012 11:48:16 -0500
Subject: [Freeswitch-dev] [Freeswitch-users] Doubts about T38 passthru
In-Reply-To: <1346258610.1241.12.camel@luna.madrid.commsmundi.com>
References: <1346229988.23367.51.camel@luna.madrid.commsmundi.com>
	<1346258610.1241.12.camel@luna.madrid.commsmundi.com>
Message-ID: <CAKbxfG-urZAjwYGuNBtfHpEEY+LMEDsd_qMHkOT_ceAB56c+qg@mail.gmail.com>

Because FS is a b2bua it probably only snaps to what type of channel
the B leg is.
FS is not a proxy and only has modest ability to emulate a few proxy behaviors.


On Wed, Aug 29, 2012 at 11:43 AM, Fran?ois Delawarde
<fdelawarde at wirelessmundi.com> wrote:
> Hi,
>
> Sent to the wrong ML, this probably is more a dev question:
>
> On Wed, 2012-08-29 at 10:46 +0200, Fran?ois Delawarde wrote:
>> Hi,
>>
>> Whenever my provider puts me on hold, it reINVITES with an SDP
>> containing both audio (sendonly) and image media like so:
>>
>> ...
>> m=audio 5320 RTP/AVP 18 3 8 0 96
>> c=IN IP4 <ip_address>
>> a=rtpmap:18 G729/8000
>> a=rtpmap:3 GSM/8000
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:0 PCMU/8000
>> a=sendonly
>> a=rtpmap:96 telephone-event/8000
>> m=image 5322 udptl t38
>> c=IN IP4 <ip_address>
>> a=T38FaxVersion:0
>> a=T38MaxBitRate:14400
>> a=T38FaxUdpEC:t38UDPRedundancy
>> a=T38FaxRateManagement:transferredTCF
>>
>>
>> Not sure if it's a non-standard SDP, but that's what I get. Now when
>> t38_passthru=true, it only forwards the "image" part to the other leg.
>> When t38_passthru=false, it only forwards the "audio" part.
>>
>> Is there a way to make t38_passthru forward the audio part as well? Or
>> should I just change to a better provider?
>
> Thanks,
> Fran?ois.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> 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



-- 
Anthony Minessale II

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

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

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


From dujinfang at gmail.com  Thu Aug 30 11:44:26 2012
From: dujinfang at gmail.com (Seven Du)
Date: Thu, 30 Aug 2012 15:44:26 +0800
Subject: [Freeswitch-dev] added mp3 codec, please review
Message-ID: <DEAB7309847D49FD99A8A3AC63572185@gmail.com>

Hi, 

To record mp4 with libmp4v2, I found the standard audio format is AAC but FS does't support, so I tried added a mp3 encoder, hopefully it works.

I have successfully record into mp4 but audio doesn't playback, videos looks fine.

Anyone can help review the code and give some input is appreciated.

http://fisheye.freeswitch.org/changelog/freeswitch.git?cs=084dac2540a15bfc0032ae89ce3e1b5660db604d

Playback Errors in VLC looks like this:

peg_audio decoder debug: emulated startcode (no startcode on following frame)
[0x10289faf0] mpeg_audio decoder debug: emulated startcode
[0x10289faf0] mpeg_audio decoder debug: emulated startcode
[0x10289faf0] mpeg_audio decoder debug: emulated startcode (no startcode on following frame)
[0x10289faf0] mpeg_audio decoder debug: emulated startcode
[0x10289faf0] mpeg_audio decoder debug: emulated startcode
[0x10289faf0] mpeg_audio decoder debug: emulated startcode (no startcode on following frame)
[0x10289faf0] mpeg_audio decoder debug: emulated startcode
[0x10289faf0] mpeg_audio decoder debug: emulated startcode (no startcode on following frame)
[0x105f10540] main audio output warning: buffer way too early (-126508), clearing queue
[0x105f10540] main audio output warning: timing screwed, stopping resampling
[0x10289faf0] mpeg_audio decoder debug: emulated startcode (no startcode on following frame)
[0x10289faf0] mpeg_audio decoder debug: emulated startcode
[0x10289faf0] mpeg_audio decoder debug: emulated startcode
[0x10289faf0] mpeg_audio decoder debug: emulated startcode
[0x10289faf0] mpeg_audio decoder debug: emulated startcode (no startcode on following frame)



I use mod_mp4v2(C) with libmp4v2 to record mp4 file, if also been interested I could share on github or check into the video-media-bug branch. But I really want to merge with mod_mp4 though which is C++ based.


@Paulo, I also CC you this message as I haven't figure out how to write hint track you input is welcome.

Thanks.

-- 
Seven Du
Sent with Sparrow (http://www.sparrowmailapp.com/?sig)

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From dujinfang at gmail.com  Thu Aug 30 13:00:57 2012
From: dujinfang at gmail.com (Seven Du)
Date: Thu, 30 Aug 2012 17:00:57 +0800
Subject: [Freeswitch-dev] question on file format codec
Message-ID: <C83F31F0204345B18F9F71B9D1778719@gmail.com>

Hi, 

If you noticed the video-media-bug branch, I managed to record a video file with the following methods

conference 3000 record /tmp/testrecord.fsv

uuid_record <uuid> start /tmp/testrecord.fsv


The first command works perfect as conference voice is L16, but the last one has bad audio I guess because it's PCMU ?.

I noticed a SWITCH_FILE_NATIVE flags which is supposed to disable transcoding but in my case in mod_fsv the flag is not set so should it transcode to L16 before writing to the fsv file?

Thanks. 

-- 
Seven Du
Sent with Sparrow (http://www.sparrowmailapp.com/?sig)

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From fdelawarde at wirelessmundi.com  Thu Aug 30 19:24:09 2012
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Thu, 30 Aug 2012 17:24:09 +0200
Subject: [Freeswitch-dev] [Freeswitch-users] Doubts about T38 passthru
In-Reply-To: <CAKbxfG-urZAjwYGuNBtfHpEEY+LMEDsd_qMHkOT_ceAB56c+qg@mail.gmail.com>
References: <1346229988.23367.51.camel@luna.madrid.commsmundi.com>
	<1346258610.1241.12.camel@luna.madrid.commsmundi.com>
	<CAKbxfG-urZAjwYGuNBtfHpEEY+LMEDsd_qMHkOT_ceAB56c+qg@mail.gmail.com>
Message-ID: <1346340249.1241.98.camel@luna.madrid.commsmundi.com>

The provider's SDP is weird anyway, like if it wanted to send me a fax
when putting me on hold (?).

My mum always said that there is nothing (even proxy things) that can't
be done with a b2bua. Was she wrong?

Fran?ois.


On Wed, 2012-08-29 at 11:48 -0500, Anthony Minessale wrote: 
> Because FS is a b2bua it probably only snaps to what type of channel
> the B leg is.
> FS is not a proxy and only has modest ability to emulate a few proxy behaviors.
> 
> 
> On Wed, Aug 29, 2012 at 11:43 AM, Fran?ois Delawarde
> <fdelawarde at wirelessmundi.com> wrote:
> > Hi,
> >
> > Sent to the wrong ML, this probably is more a dev question:
> >
> > On Wed, 2012-08-29 at 10:46 +0200, Fran?ois Delawarde wrote:
> >> Hi,
> >>
> >> Whenever my provider puts me on hold, it reINVITES with an SDP
> >> containing both audio (sendonly) and image media like so:
> >>
> >> ...
> >> m=audio 5320 RTP/AVP 18 3 8 0 96
> >> c=IN IP4 <ip_address>
> >> a=rtpmap:18 G729/8000
> >> a=rtpmap:3 GSM/8000
> >> a=rtpmap:8 PCMA/8000
> >> a=rtpmap:0 PCMU/8000
> >> a=sendonly
> >> a=rtpmap:96 telephone-event/8000
> >> m=image 5322 udptl t38
> >> c=IN IP4 <ip_address>
> >> a=T38FaxVersion:0
> >> a=T38MaxBitRate:14400
> >> a=T38FaxUdpEC:t38UDPRedundancy
> >> a=T38FaxRateManagement:transferredTCF
> >>
> >>
> >> Not sure if it's a non-standard SDP, but that's what I get. Now when
> >> t38_passthru=true, it only forwards the "image" part to the other leg.
> >> When t38_passthru=false, it only forwards the "audio" part.
> >>
> >> Is there a way to make t38_passthru forward the audio part as well? Or
> >> should I just change to a better provider?
> >
> > Thanks,
> > Fran?ois.



From anthony.minessale at gmail.com  Thu Aug 30 19:31:44 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 30 Aug 2012 10:31:44 -0500
Subject: [Freeswitch-dev] [Freeswitch-users] Doubts about T38 passthru
In-Reply-To: <1346340249.1241.98.camel@luna.madrid.commsmundi.com>
References: <1346229988.23367.51.camel@luna.madrid.commsmundi.com>
	<1346258610.1241.12.camel@luna.madrid.commsmundi.com>
	<CAKbxfG-urZAjwYGuNBtfHpEEY+LMEDsd_qMHkOT_ceAB56c+qg@mail.gmail.com>
	<1346340249.1241.98.camel@luna.madrid.commsmundi.com>
Message-ID: <CAKbxfG_Z3Em-hdFtJVipZUnbSvXPnSE-EpoPbqHw_FY8TZunQg@mail.gmail.com>

indeed she is, b2bua is 2 separate dialogs where proxy has the
connivence of forwarding 1 dialog.


On Thu, Aug 30, 2012 at 10:24 AM, Fran?ois Delawarde
<fdelawarde at wirelessmundi.com> wrote:
> The provider's SDP is weird anyway, like if it wanted to send me a fax
> when putting me on hold (?).
>
> My mum always said that there is nothing (even proxy things) that can't
> be done with a b2bua. Was she wrong?
>
> Fran?ois.
>
>
> On Wed, 2012-08-29 at 11:48 -0500, Anthony Minessale wrote:
>> Because FS is a b2bua it probably only snaps to what type of channel
>> the B leg is.
>> FS is not a proxy and only has modest ability to emulate a few proxy behaviors.
>>
>>
>> On Wed, Aug 29, 2012 at 11:43 AM, Fran?ois Delawarde
>> <fdelawarde at wirelessmundi.com> wrote:
>> > Hi,
>> >
>> > Sent to the wrong ML, this probably is more a dev question:
>> >
>> > On Wed, 2012-08-29 at 10:46 +0200, Fran?ois Delawarde wrote:
>> >> Hi,
>> >>
>> >> Whenever my provider puts me on hold, it reINVITES with an SDP
>> >> containing both audio (sendonly) and image media like so:
>> >>
>> >> ...
>> >> m=audio 5320 RTP/AVP 18 3 8 0 96
>> >> c=IN IP4 <ip_address>
>> >> a=rtpmap:18 G729/8000
>> >> a=rtpmap:3 GSM/8000
>> >> a=rtpmap:8 PCMA/8000
>> >> a=rtpmap:0 PCMU/8000
>> >> a=sendonly
>> >> a=rtpmap:96 telephone-event/8000
>> >> m=image 5322 udptl t38
>> >> c=IN IP4 <ip_address>
>> >> a=T38FaxVersion:0
>> >> a=T38MaxBitRate:14400
>> >> a=T38FaxUdpEC:t38UDPRedundancy
>> >> a=T38FaxRateManagement:transferredTCF
>> >>
>> >>
>> >> Not sure if it's a non-standard SDP, but that's what I get. Now when
>> >> t38_passthru=true, it only forwards the "image" part to the other leg.
>> >> When t38_passthru=false, it only forwards the "audio" part.
>> >>
>> >> Is there a way to make t38_passthru forward the audio part as well? Or
>> >> should I just change to a better provider?
>> >
>> > Thanks,
>> > Fran?ois.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> 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



-- 
Anthony Minessale II

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

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

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


From fdelawarde at wirelessmundi.com  Thu Aug 30 20:09:28 2012
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Thu, 30 Aug 2012 18:09:28 +0200
Subject: [Freeswitch-dev] [Freeswitch-users] Doubts about T38 passthru
In-Reply-To: <CAKbxfG_Z3Em-hdFtJVipZUnbSvXPnSE-EpoPbqHw_FY8TZunQg@mail.gmail.com>
References: <1346229988.23367.51.camel@luna.madrid.commsmundi.com>
	<1346258610.1241.12.camel@luna.madrid.commsmundi.com>
	<CAKbxfG-urZAjwYGuNBtfHpEEY+LMEDsd_qMHkOT_ceAB56c+qg@mail.gmail.com>
	<1346340249.1241.98.camel@luna.madrid.commsmundi.com>
	<CAKbxfG_Z3Em-hdFtJVipZUnbSvXPnSE-EpoPbqHw_FY8TZunQg@mail.gmail.com>
Message-ID: <1346342968.1241.121.camel@luna.madrid.commsmundi.com>

I don't see what you can't do with a b2bua in terms of features compared
to a proxy, apart from it (the proxy) being faster and more reliable.


On Thu, 2012-08-30 at 10:31 -0500, Anthony Minessale wrote: 
> indeed she is, b2bua is 2 separate dialogs where proxy has the
> connivence of forwarding 1 dialog.
> 
> 
> On Thu, Aug 30, 2012 at 10:24 AM, Fran?ois Delawarde
> <fdelawarde at wirelessmundi.com> wrote:
> > The provider's SDP is weird anyway, like if it wanted to send me a fax
> > when putting me on hold (?).
> >
> > My mum always said that there is nothing (even proxy things) that can't
> > be done with a b2bua. Was she wrong?
> >
> > Fran?ois.
> >
> >
> > On Wed, 2012-08-29 at 11:48 -0500, Anthony Minessale wrote:
> >> Because FS is a b2bua it probably only snaps to what type of channel
> >> the B leg is.
> >> FS is not a proxy and only has modest ability to emulate a few proxy behaviors.
> >>
> >>
> >> On Wed, Aug 29, 2012 at 11:43 AM, Fran?ois Delawarde
> >> <fdelawarde at wirelessmundi.com> wrote:
> >> > Hi,
> >> >
> >> > Sent to the wrong ML, this probably is more a dev question:
> >> >
> >> > On Wed, 2012-08-29 at 10:46 +0200, Fran?ois Delawarde wrote:
> >> >> Hi,
> >> >>
> >> >> Whenever my provider puts me on hold, it reINVITES with an SDP
> >> >> containing both audio (sendonly) and image media like so:
> >> >>
> >> >> ...
> >> >> m=audio 5320 RTP/AVP 18 3 8 0 96
> >> >> c=IN IP4 <ip_address>
> >> >> a=rtpmap:18 G729/8000
> >> >> a=rtpmap:3 GSM/8000
> >> >> a=rtpmap:8 PCMA/8000
> >> >> a=rtpmap:0 PCMU/8000
> >> >> a=sendonly
> >> >> a=rtpmap:96 telephone-event/8000
> >> >> m=image 5322 udptl t38
> >> >> c=IN IP4 <ip_address>
> >> >> a=T38FaxVersion:0
> >> >> a=T38MaxBitRate:14400
> >> >> a=T38FaxUdpEC:t38UDPRedundancy
> >> >> a=T38FaxRateManagement:transferredTCF
> >> >>
> >> >>
> >> >> Not sure if it's a non-standard SDP, but that's what I get. Now when
> >> >> t38_passthru=true, it only forwards the "image" part to the other leg.
> >> >> When t38_passthru=false, it only forwards the "audio" part.
> >> >>
> >> >> Is there a way to make t38_passthru forward the audio part as well? Or
> >> >> should I just change to a better provider?
> >> >
> >> > Thanks,
> >> > Fran?ois.
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > 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 anthony.minessale at gmail.com  Fri Aug 31 00:36:14 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 30 Aug 2012 15:36:14 -0500
Subject: [Freeswitch-dev] question on file format codec
In-Reply-To: <C83F31F0204345B18F9F71B9D1778719@gmail.com>
References: <C83F31F0204345B18F9F71B9D1778719@gmail.com>
Message-ID: <CAKbxfG_RoRR0fcFTuVRKR3T9v4-5kzhYFEYCUrQA1BmQrf3nuw@mail.gmail.com>

yes, currently the fsv format expects L16 audio

On Thu, Aug 30, 2012 at 4:00 AM, Seven Du <dujinfang at gmail.com> wrote:
> Hi,
>
> If you noticed the video-media-bug branch, I managed to record a video file
> with the following methods
>
> conference 3000 record /tmp/testrecord.fsv
>
> uuid_record <uuid> start /tmp/testrecord.fsv
>
>
> The first command works perfect as conference voice is L16, but the last one
> has bad audio I guess because it's PCMU ?.
>
> I noticed a SWITCH_FILE_NATIVE flags which is supposed to disable
> transcoding but in my case in mod_fsv the flag is not set so should it
> transcode to L16 before writing to the fsv file?
>
> Thanks.
>
> --
> Seven Du
> Sent with Sparrow
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> 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
>



-- 
Anthony Minessale II

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

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

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


From anthony.minessale at gmail.com  Fri Aug 31 00:41:13 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 30 Aug 2012 15:41:13 -0500
Subject: [Freeswitch-dev] [Freeswitch-users] Doubts about T38 passthru
In-Reply-To: <1346342968.1241.121.camel@luna.madrid.commsmundi.com>
References: <1346229988.23367.51.camel@luna.madrid.commsmundi.com>
	<1346258610.1241.12.camel@luna.madrid.commsmundi.com>
	<CAKbxfG-urZAjwYGuNBtfHpEEY+LMEDsd_qMHkOT_ceAB56c+qg@mail.gmail.com>
	<1346340249.1241.98.camel@luna.madrid.commsmundi.com>
	<CAKbxfG_Z3Em-hdFtJVipZUnbSvXPnSE-EpoPbqHw_FY8TZunQg@mail.gmail.com>
	<1346342968.1241.121.camel@luna.madrid.commsmundi.com>
Message-ID: <CAKbxfG9ktCK4cmg-gS4cJ8qhMH+vW-gouOgMKYKR=xthXkhO9g@mail.gmail.com>

As long as you understand the difference between the fact that
messages passing across a bridge on a b2bua is much more complex than
on a proxy and will not match up 100%

On Thu, Aug 30, 2012 at 11:09 AM, Fran?ois Delawarde
<fdelawarde at wirelessmundi.com> wrote:
> I don't see what you can't do with a b2bua in terms of features compared
> to a proxy, apart from it (the proxy) being faster and more reliable.
>
>
> On Thu, 2012-08-30 at 10:31 -0500, Anthony Minessale wrote:
>> indeed she is, b2bua is 2 separate dialogs where proxy has the
>> connivence of forwarding 1 dialog.
>>
>>
>> On Thu, Aug 30, 2012 at 10:24 AM, Fran?ois Delawarde
>> <fdelawarde at wirelessmundi.com> wrote:
>> > The provider's SDP is weird anyway, like if it wanted to send me a fax
>> > when putting me on hold (?).
>> >
>> > My mum always said that there is nothing (even proxy things) that can't
>> > be done with a b2bua. Was she wrong?
>> >
>> > Fran?ois.
>> >
>> >
>> > On Wed, 2012-08-29 at 11:48 -0500, Anthony Minessale wrote:
>> >> Because FS is a b2bua it probably only snaps to what type of channel
>> >> the B leg is.
>> >> FS is not a proxy and only has modest ability to emulate a few proxy behaviors.
>> >>
>> >>
>> >> On Wed, Aug 29, 2012 at 11:43 AM, Fran?ois Delawarde
>> >> <fdelawarde at wirelessmundi.com> wrote:
>> >> > Hi,
>> >> >
>> >> > Sent to the wrong ML, this probably is more a dev question:
>> >> >
>> >> > On Wed, 2012-08-29 at 10:46 +0200, Fran?ois Delawarde wrote:
>> >> >> Hi,
>> >> >>
>> >> >> Whenever my provider puts me on hold, it reINVITES with an SDP
>> >> >> containing both audio (sendonly) and image media like so:
>> >> >>
>> >> >> ...
>> >> >> m=audio 5320 RTP/AVP 18 3 8 0 96
>> >> >> c=IN IP4 <ip_address>
>> >> >> a=rtpmap:18 G729/8000
>> >> >> a=rtpmap:3 GSM/8000
>> >> >> a=rtpmap:8 PCMA/8000
>> >> >> a=rtpmap:0 PCMU/8000
>> >> >> a=sendonly
>> >> >> a=rtpmap:96 telephone-event/8000
>> >> >> m=image 5322 udptl t38
>> >> >> c=IN IP4 <ip_address>
>> >> >> a=T38FaxVersion:0
>> >> >> a=T38MaxBitRate:14400
>> >> >> a=T38FaxUdpEC:t38UDPRedundancy
>> >> >> a=T38FaxRateManagement:transferredTCF
>> >> >>
>> >> >>
>> >> >> Not sure if it's a non-standard SDP, but that's what I get. Now when
>> >> >> t38_passthru=true, it only forwards the "image" part to the other leg.
>> >> >> When t38_passthru=false, it only forwards the "audio" part.
>> >> >>
>> >> >> Is there a way to make t38_passthru forward the audio part as well? Or
>> >> >> should I just change to a better provider?
>> >> >
>> >> > Thanks,
>> >> > Fran?ois.
>> >
>> >
>> > _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > 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
>>
>>
>>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> 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



-- 
Anthony Minessale II

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

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

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


From dujinfang at gmail.com  Fri Aug 31 04:22:42 2012
From: dujinfang at gmail.com (dujinfang)
Date: Fri, 31 Aug 2012 08:22:42 +0800
Subject: [Freeswitch-dev] question on file format codec
In-Reply-To: <CAKbxfG_RoRR0fcFTuVRKR3T9v4-5kzhYFEYCUrQA1BmQrf3nuw@mail.gmail.com>
References: <C83F31F0204345B18F9F71B9D1778719@gmail.com>
	<CAKbxfG_RoRR0fcFTuVRKR3T9v4-5kzhYFEYCUrQA1BmQrf3nuw@mail.gmail.com>
Message-ID: <735FEE21-1A3B-4FF3-9DD8-2BD4AFE24DE7@gmail.com>

yes, my problem is how to make sure switch_file
_write always get L16 data when trigger by uuid_record?

thanks

???? iPhone

? 2012-8-31???4:36?Anthony Minessale <anthony.minessale at gmail.com> ???

> yes, currently the fsv format expects L16 audio
> 
> On Thu, Aug 30, 2012 at 4:00 AM, Seven Du <dujinfang at gmail.com> wrote:
>> Hi,
>> 
>> If you noticed the video-media-bug branch, I managed to record a video file
>> with the following methods
>> 
>> conference 3000 record /tmp/testrecord.fsv
>> 
>> uuid_record <uuid> start /tmp/testrecord.fsv
>> 
>> 
>> The first command works perfect as conference voice is L16, but the last one
>> has bad audio I guess because it's PCMU ?.
>> 
>> I noticed a SWITCH_FILE_NATIVE flags which is supposed to disable
>> transcoding but in my case in mod_fsv the flag is not set so should it
>> transcode to L16 before writing to the fsv file?
>> 
>> Thanks.
>> 
>> --
>> Seven Du
>> Sent with Sparrow
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> 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
>> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-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 anthony.minessale at gmail.com  Fri Aug 31 04:34:29 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 30 Aug 2012 19:34:29 -0500
Subject: [Freeswitch-dev] question on file format codec
In-Reply-To: <735FEE21-1A3B-4FF3-9DD8-2BD4AFE24DE7@gmail.com>
References: <C83F31F0204345B18F9F71B9D1778719@gmail.com>
	<CAKbxfG_RoRR0fcFTuVRKR3T9v4-5kzhYFEYCUrQA1BmQrf3nuw@mail.gmail.com>
	<735FEE21-1A3B-4FF3-9DD8-2BD4AFE24DE7@gmail.com>
Message-ID: <CAKbxfG8jx+oKT4DeezxH9ngSgDmx951GQbugT-r6bxkAQJSaEQ@mail.gmail.com>

Right now it always does raw l16 in any of the io or file funcs
On Aug 30, 2012 7:23 PM, "dujinfang" <dujinfang at gmail.com> wrote:

> yes, my problem is how to make sure switch_file
> _write always get L16 data when trigger by uuid_record?
>
> thanks
>
> ???? iPhone
>
> ? 2012-8-31???4:36?Anthony Minessale <anthony.minessale at gmail.com> ???
>
> > yes, currently the fsv format expects L16 audio
> >
> > On Thu, Aug 30, 2012 at 4:00 AM, Seven Du <dujinfang at gmail.com> wrote:
> >> Hi,
> >>
> >> If you noticed the video-media-bug branch, I managed to record a video
> file
> >> with the following methods
> >>
> >> conference 3000 record /tmp/testrecord.fsv
> >>
> >> uuid_record <uuid> start /tmp/testrecord.fsv
> >>
> >>
> >> The first command works perfect as conference voice is L16, but the
> last one
> >> has bad audio I guess because it's PCMU ?.
> >>
> >> I noticed a SWITCH_FILE_NATIVE flags which is supposed to disable
> >> transcoding but in my case in mod_fsv the flag is not set so should it
> >> transcode to L16 before writing to the fsv file?
> >>
> >> Thanks.
> >>
> >> --
> >> Seven Du
> >> Sent with Sparrow
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.com
> >>
> >> 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
> >>
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org
> > googletalk:conf+888 at conference.freeswitch.org
> > pstn:+19193869900
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> 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
>
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From dujinfang at gmail.com  Fri Aug 31 07:15:46 2012
From: dujinfang at gmail.com (Seven Du)
Date: Fri, 31 Aug 2012 11:15:46 +0800
Subject: [Freeswitch-dev] question on file format codec
In-Reply-To: <CAKbxfG8jx+oKT4DeezxH9ngSgDmx951GQbugT-r6bxkAQJSaEQ@mail.gmail.com>
References: <C83F31F0204345B18F9F71B9D1778719@gmail.com>
	<CAKbxfG_RoRR0fcFTuVRKR3T9v4-5kzhYFEYCUrQA1BmQrf3nuw@mail.gmail.com>
	<735FEE21-1A3B-4FF3-9DD8-2BD4AFE24DE7@gmail.com>
	<CAKbxfG8jx+oKT4DeezxH9ngSgDmx951GQbugT-r6bxkAQJSaEQ@mail.gmail.com>
Message-ID: <2E54E1470F48416CB1CD7FE338C6340A@gmail.com>

ok, I figured out when use conference record, handle->channels = 1, but in uuid_record, handle->channels = 2, so the following code did the trick in fsv_file_write, but I'm not sure if it's the write approach. Why the *len is always 160 in both cases? where the other half channel data goes to? I use play_fsv so the playback should be right. Only tested on 8000HZ right now. If the following code is confusing I can submit a full patch. Thanks.  



uint16_t *x = data;


if (handle->channels == 2) {
/* should we use only one channel ? */
for (int i=0; i<160; i++) {
x[i] = x[i*2];
}
}



--  
Seven Du
Sent with Sparrow (http://www.sparrowmailapp.com/?sig)


On Friday, August 31, 2012 at 8:34 AM, Anthony Minessale wrote:

> Right now it always does raw l16 in any of the io or file funcs
> On Aug 30, 2012 7:23 PM, "dujinfang" <dujinfang at gmail.com (mailto:dujinfang at gmail.com)> wrote:
> > yes, my problem is how to make sure switch_file
> > _write always get L16 data when trigger by uuid_record?
> >  
> > thanks
> >  
> > ???? iPhone
> >  
> > ? 2012-8-31???4:36?Anthony Minessale <anthony.minessale at gmail.com (mailto:anthony.minessale at gmail.com)> ???
> >  
> > > yes, currently the fsv format expects L16 audio
> > >
> > > On Thu, Aug 30, 2012 at 4:00 AM, Seven Du <dujinfang at gmail.com (mailto:dujinfang at gmail.com)> wrote:
> > >> Hi,
> > >>
> > >> If you noticed the video-media-bug branch, I managed to record a video file
> > >> with the following methods
> > >>
> > >> conference 3000 record /tmp/testrecord.fsv
> > >>
> > >> uuid_record <uuid> start /tmp/testrecord.fsv
> > >>
> > >>
> > >> The first command works perfect as conference voice is L16, but the last one
> > >> has bad audio I guess because it's PCMU ?.
> > >>
> > >> I noticed a SWITCH_FILE_NATIVE flags which is supposed to disable
> > >> transcoding but in my case in mod_fsv the flag is not set so should it
> > >> transcode to L16 before writing to the fsv file?
> > >>
> > >> Thanks.
> > >>
> > >> --
> > >> Seven Du
> > >> Sent with Sparrow
> > >>
> > >>
> > >> _________________________________________________________________________
> > >> Professional FreeSWITCH Consulting Services:
> > >> consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> > >> http://www.freeswitchsolutions.com
> > >>
> > >> 
> > >> 
> > >>
> > >> Official FreeSWITCH Sites
> > >> http://www.freeswitch.org
> > >> http://wiki.freeswitch.org
> > >> http://www.cluecon.com
> > >>
> > >> FreeSWITCH-dev mailing list
> > >> FreeSWITCH-dev at lists.freeswitch.org (mailto: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
> > >>
> > >
> > >
> > >
> > > --
> > > Anthony Minessale II
> > >
> > > FreeSWITCH http://www.freeswitch.org/
> > > ClueCon http://www.cluecon.com/
> > > Twitter: http://twitter.com/FreeSWITCH_wire
> > >
> > > AIM: anthm
> > > MSN:anthony_minessale at hotmail.com (mailto:MSN%3Aanthony_minessale at hotmail.com)
> > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com (mailto:PAYPAL%3Aanthony.minessale at gmail.com)
> > > IRC: irc.freenode.net (http://irc.freenode.net) #freeswitch
> > >
> > > FreeSWITCH Developer Conference
> > > sip:888 at conference.freeswitch.org (mailto:sip%3A888 at conference.freeswitch.org)
> > > googletalk:conf+888 at conference.freeswitch.org (mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org)
> > > pstn:+19193869900 (tel:%2B19193869900)
> > >
> > > _________________________________________________________________________
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> > > http://www.freeswitchsolutions.com
> > >
> > > 
> > > 
> > >
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://wiki.freeswitch.org
> > > http://www.cluecon.com
> > >
> > > FreeSWITCH-dev mailing list
> > > FreeSWITCH-dev at lists.freeswitch.org (mailto: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
> >  
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> > http://www.freeswitchsolutions.com
> >  
> > 
> > 
> >  
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >  
> > FreeSWITCH-dev mailing list
> > FreeSWITCH-dev at lists.freeswitch.org (mailto: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
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> http://www.freeswitchsolutions.com
>  
> 
> 
>  
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>  
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org (mailto: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
>  
>  


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From anthony.minessale at gmail.com  Fri Aug 31 21:28:11 2012
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 31 Aug 2012 12:28:11 -0500
Subject: [Freeswitch-dev] question on file format codec
In-Reply-To: <2E54E1470F48416CB1CD7FE338C6340A@gmail.com>
References: <C83F31F0204345B18F9F71B9D1778719@gmail.com>
	<CAKbxfG_RoRR0fcFTuVRKR3T9v4-5kzhYFEYCUrQA1BmQrf3nuw@mail.gmail.com>
	<735FEE21-1A3B-4FF3-9DD8-2BD4AFE24DE7@gmail.com>
	<CAKbxfG8jx+oKT4DeezxH9ngSgDmx951GQbugT-r6bxkAQJSaEQ@mail.gmail.com>
	<2E54E1470F48416CB1CD7FE338C6340A@gmail.com>
Message-ID: <CAKbxfG9e0BywCNhaSjam+cqu0gcfgcF1wTZv7KXdM6daMTTTLA@mail.gmail.com>

when you have multiple channels it still only uses the len of one
channel assuming you will multiply that len by the * of channels for
the real size.


On Thu, Aug 30, 2012 at 10:15 PM, Seven Du <dujinfang at gmail.com> wrote:
> ok, I figured out when use conference record, handle->channels = 1, but in
> uuid_record, handle->channels = 2, so the following code did the trick in
> fsv_file_write, but I'm not sure if it's the write approach. Why the *len is
> always 160 in both cases? where the other half channel data goes to? I use
> play_fsv so the playback should be right. Only tested on 8000HZ right now.
> If the following code is confusing I can submit a full patch. Thanks.
>
>
>
> uint16_t *x = data;
>
>
> if (handle->channels == 2) {
> /* should we use only one channel ? */
> for (int i=0; i<160; i++) {
> x[i] = x[i*2];
> }
> }
>
>
> --
> Seven Du
> Sent with Sparrow
>
> On Friday, August 31, 2012 at 8:34 AM, Anthony Minessale wrote:
>
> Right now it always does raw l16 in any of the io or file funcs
>
> On Aug 30, 2012 7:23 PM, "dujinfang" <dujinfang at gmail.com> wrote:
>
> yes, my problem is how to make sure switch_file
> _write always get L16 data when trigger by uuid_record?
>
> thanks
>
> ???? iPhone
>
> ? 2012-8-31???4:36?Anthony Minessale <anthony.minessale at gmail.com> ???
>
>> yes, currently the fsv format expects L16 audio
>>
>> On Thu, Aug 30, 2012 at 4:00 AM, Seven Du <dujinfang at gmail.com> wrote:
>>> Hi,
>>>
>>> If you noticed the video-media-bug branch, I managed to record a video
>>> file
>>> with the following methods
>>>
>>> conference 3000 record /tmp/testrecord.fsv
>>>
>>> uuid_record <uuid> start /tmp/testrecord.fsv
>>>
>>>
>>> The first command works perfect as conference voice is L16, but the last
>>> one
>>> has bad audio I guess because it's PCMU ?.
>>>
>>> I noticed a SWITCH_FILE_NATIVE flags which is supposed to disable
>>> transcoding but in my case in mod_fsv the flag is not set so should it
>>> transcode to L16 before writing to the fsv file?
>>>
>>> Thanks.
>>>
>>> --
>>> Seven Du
>>> Sent with Sparrow
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> 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
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>>
>> FreeSWITCH-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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> 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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> 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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> 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
>



-- 
Anthony Minessale II

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

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

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