From msc at freeswitch.org  Sat Oct  1 00:57:56 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 30 Sep 2011 13:57:56 -0700
Subject: [Freeswitch-dev] FreeSWITCH Conference Call Oct 5: All about git
Message-ID: <CAKzWOxVHZfYM97CTko=KdgEf-=xWHH0hsX4wEjEaN1JCMg03Jw@mail.gmail.com>

Hello FreeSWITCHers!

Travis Cross will be presenting on the subject of git. Many of us have a
love-hate (or a hate-hate) relationship with git so Travis is going to help
us get over the hump. The agenda page has a place where you can post
questions:

http://wiki.freeswitch.org/wiki/FS_weekly_2011_10_05#Featured_Presentation

Looking forward to next Wednesday!

-MC
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From succer110 at tiscali.it  Mon Oct  3 16:24:28 2011
From: succer110 at tiscali.it (succer110 at tiscali.it)
Date: Mon, 3 Oct 2011 14:24:28 +0200 (CEST)
Subject: [Freeswitch-dev] memory pool
Message-ID: <11030571.102811317644668479.JavaMail.defaultUser@defaultHost>

Hi! I'm a new fs module developer :)
In the page "authoring freeswitch module" i read that "there's still times when using traditional C memory management is a better plan."
How can i recognize that cases?

>From what i have understood using memory pool from apr i can forget to call switch_core_destroy_memory_pool unless i specifically want it. Am i Right? Is possibile to specify the dimension of the memory pool i want to create? Which is the standard dimension?
Thankyou =)

E' nata indoona: chiama, videochiama e messaggia Gratis. 
Scarica indoona per iPhone, Android e PC

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From anthony.minessale at gmail.com  Mon Oct  3 17:39:52 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 3 Oct 2011 08:39:52 -0500
Subject: [Freeswitch-dev] memory pool
In-Reply-To: <11030571.102811317644668479.JavaMail.defaultUser@defaultHost>
References: <11030571.102811317644668479.JavaMail.defaultUser@defaultHost>
Message-ID: <CAKbxfG8aNAMqrK3tVH3AhRy7zUbNvfx6=qtROfw-7RzxKakpOw@mail.gmail.com>

On Mon, Oct 3, 2011 at 7:24 AM, succer110 at tiscali.it
<succer110 at tiscali.it> wrote:
> Hi! I'm a new fs module developer :)
> In the page "authoring freeswitch module" i read that "there's still times
> when using traditional C memory management is a better plan."
> How can i recognize that cases?

If you do not have a well-defined lifecycle of an object or need
several temporary allocations repeatedly its better to use malloc.
Pools are best used when you can create and destroy the pool regularly
based on the workflow of you application such as a main object
representing an handle etc.

Anything using a session can also use the session's pool.

>
> From what i have understood using memory pool from apr i can forget to call
> switch_core_destroy_memory_pool unless i specifically want it. Am i Right?
> Is possibile to specify the dimension of the memory pool i want to create?
> Which is the standard dimension?
> Thankyou =)
>

The pools auto-size themselves and do not support freeing any
allocated data.  The whole pool must be destroyed.


> E' nata indoona: chiama, videochiama e messaggia Gratis.
> Scarica indoona per iPhone, Android e PC
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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 juraj.fabo at gmail.com  Tue Oct  4 16:56:17 2011
From: juraj.fabo at gmail.com (Juraj Fabo)
Date: Tue, 4 Oct 2011 14:56:17 +0200
Subject: [Freeswitch-dev] MTU setting and application buffer size
Message-ID: <CAAqVCyTaPsWz2tmShrpEgwFmY27nFaw0AOHDq3Mmn932QTQ4aQ@mail.gmail.com>

Hi,

I read through the previous answers several times again and did some progress.
I found out that the valid codec_ms values are <10ms,60ms> and using lower
values will not work.

>> 80 means a hardware interrupt is received every 10ms with 80 bytes per
>> time slot. This is the recommended mode. A value of 40 will increase
>> interrupt load and not necesarily reduce your latency, you must reduce
>> the "user period", which is how often the driver will deliver
>> media/data to the user application. This is done using
>> sangoma_tdm_set_usr_period().
>>
>>
>> The codec_ms is used to call sangoma_tdm_set_usr_period(). This is how
>> often the driver will deliver data to the user application (for
>> example, waking it up from select()).
>>

In each of the the test below the codec_ms was set to 1 with hope
of fastest data delivery, however this resulted to 20ms codec_ms
which is default.


> Please, have a look at the results with various parameters used:
> mtu:80 txqueue_size=1 ?rxqueue_size=1 ? one-direction delay=60ms
> round-trip delay=120ms
> mtu:80 txqueue_size=2 ?rxqueue_size=2 ? one-direction delay=80ms
> round-trip delay=160ms
> mtu:80 txqueue_size=10 rxqueue_size=10 one-direction delay=240ms
> round-trip delay=480ms
> mtu:40 txqueue_size=1 ?rxqueue_size=1 ? one-direction delay=40ms
> round-trip delay=80ms
> mtu:40 txqueue_size=2 ?rxqueue_size=2 ? one-direction delay=60ms
> round-trip delay=120ms
> mtu:40 txqueue_size=5 ?rxqueue_size=5 ? one-direction delay=120ms
> round-trip delay=240ms
> mtu:16 txqueue_size=1 ?rxqueue_size=1 ? one-direction delay=30ms
> round-trip delay=60ms
> mtu:8 ?txqueue_size=1 ? rxqueue_size=1 ? one-direction delay=25ms
> round-trip delay=50ms
> mtu:8 ?txqueue_size=10 rxqueue_size=10 one-direction delay=205ms
> round-trip delay=410ms
>

Updated test result with the codec_ms=10 is following:
mtu:16 txqueue_size=1  rxqueue_size=10   one-direction delay=20ms
round-trip delay=50ms
mtu:80 txqueue_size=1  rxqueue_size=10   one-direction delay=56ms
round-trip delay=110ms

However, this is still not 10ms with the default MTU:80 and codec_ms:10 :(

Anyway, is it a good idea to use so assymetric tx/rx queue sizes?

with best regards

Juraj


From msc at freeswitch.org  Wed Oct  5 20:19:44 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Oct 2011 09:19:44 -0700
Subject: [Freeswitch-dev] FreeSWITCH Conference Call Today: Git Tutorial
Message-ID: <CAKzWOxXkdLmDJwpa=fOL7ZDvRXT+rGs3tQPeQTRkX__b_iQgxA@mail.gmail.com>

Hello all!

Today we have our very own Travis Cross who will be helping everyone get up
to speed with git. We highly recommend everyone who uses the FreeSWITCH
source to join us and get your git questions answered. The agenda page is
here:

http://wiki.freeswitch.org/wiki/FS_weekly_2011_10_05

Talk to you soon,
Michael
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From gmaruzz at gmail.com  Wed Oct  5 21:00:52 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 5 Oct 2011 19:00:52 +0200
Subject: [Freeswitch-dev] FreeSWITCH Conference Call Today: Git Tutorial
In-Reply-To: <CAKzWOxXkdLmDJwpa=fOL7ZDvRXT+rGs3tQPeQTRkX__b_iQgxA@mail.gmail.com>
References: <CAKzWOxXkdLmDJwpa=fOL7ZDvRXT+rGs3tQPeQTRkX__b_iQgxA@mail.gmail.com>
Message-ID: <CALXCt0o9JF0OCSZ55D5J6Yt_g5_Mw8PczzPfU5wY9zz7OtGbMg@mail.gmail.com>

I'll try to join from cell, but please please please, do a recording anyway ;).

-giovanni


On 10/5/11, Michael Collins <msc at freeswitch.org> wrote:
> Hello all!
>
> Today we have our very own Travis Cross who will be helping everyone get up
> to speed with git. We highly recommend everyone who uses the FreeSWITCH
> source to join us and get your git questions answered. The agenda page is
> here:
>
> http://wiki.freeswitch.org/wiki/FS_weekly_2011_10_05
>
> Talk to you soon,
> Michael
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From xin at ind.rwth-aachen.de  Tue Oct  4 16:03:51 2011
From: xin at ind.rwth-aachen.de (Han Xin)
Date: Tue, 04 Oct 2011 14:03:51 +0200
Subject: [Freeswitch-dev] trouble on how to remove the media bug from the
 session after the sessions are closed
Message-ID: <94715a0a3f69399835a066ea4d8ea62c@gw.ind.rwth-aachen.de>

Hello everyone!

I am new to Freeswitch and try to add media bugs such as digital signal processing to the read stream and write stream. But I have some trouble on how to remove the media bug from the session after the sessions are closed. I tried to look into other existed modules such as mod_soundtouch and mod_dptools to find a solution, the former mod_soundtouch put the function "switch_core_media_bug_remove()" in the beginning of application run function. The latter mod use a single app called "remove bugs" and put this app in the dial plan to remove the bugs.

The question is where should I put this "switch_core_media_bug_remove()" function after the sessions are closed.
Is there a way to automatically remove a media bug when a call is ended?
Or it is only possible to use a single app to remove the media bugs as in the mod_dptools?

Thank you all in advance!

--





From vutamhoan at gmail.com  Wed Oct  5 04:46:57 2011
From: vutamhoan at gmail.com (Vu Quang Hoa)
Date: Wed, 5 Oct 2011 07:46:57 +0700
Subject: [Freeswitch-dev] MOH in async mode is possible?
Message-ID: <CAJnMD2cRRu8FC_W0Jt5hyPmourSxG=X+6a7R5eVga1V6mZZxbw@mail.gmail.com>

Hi everyone,

I've been trying ways to provide MOH in background of Lua script (searching
and found few way) but none of them really satisfied me. I think it
may implement in the same way of record_session function. So I need
your advises if it is possible or not? It's a really good idea?

Thank you!
vutamhoan
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From cmrienzo at gmail.com  Thu Oct  6 04:35:48 2011
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Wed, 5 Oct 2011 20:35:48 -0400
Subject: [Freeswitch-dev] trouble on how to remove the media bug from
 the session after the sessions are closed
In-Reply-To: <94715a0a3f69399835a066ea4d8ea62c@gw.ind.rwth-aachen.de>
References: <94715a0a3f69399835a066ea4d8ea62c@gw.ind.rwth-aachen.de>
Message-ID: <CAD-qRkjwVNoe4f89WGYM-jCFHXvA3ve8Q9bjfp8Hb0iC=LQN+A@mail.gmail.com>

Media bugs are automatically removed when the call ends.  If you need to
remove a media bug at an earlier point in the call, you can use
switch_core_media_bug_remove(session, bug).



On Tue, Oct 4, 2011 at 8:03 AM, Han Xin <xin at ind.rwth-aachen.de> wrote:

> Hello everyone!
>
> I am new to Freeswitch and try to add media bugs such as digital signal
> processing to the read stream and write stream. But I have some trouble on
> how to remove the media bug from the session after the sessions are closed.
> I tried to look into other existed modules such as mod_soundtouch and
> mod_dptools to find a solution, the former mod_soundtouch put the function
> "switch_core_media_bug_remove()" in the beginning of application run
> function. The latter mod use a single app called "remove bugs" and put this
> app in the dial plan to remove the bugs.
>
> The question is where should I put this "switch_core_media_bug_remove()"
> function after the sessions are closed.
> Is there a way to automatically remove a media bug when a call is ended?
> Or it is only possible to use a single app to remove the media bugs as in
> the mod_dptools?
>
> Thank you all in advance!
>
> --
>
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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 anthony.minessale at gmail.com  Thu Oct  6 19:08:47 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 6 Oct 2011 10:08:47 -0500
Subject: [Freeswitch-dev] MOH in async mode is possible?
In-Reply-To: <CAJnMD2cRRu8FC_W0Jt5hyPmourSxG=X+6a7R5eVga1V6mZZxbw@mail.gmail.com>
References: <CAJnMD2cRRu8FC_W0Jt5hyPmourSxG=X+6a7R5eVga1V6mZZxbw@mail.gmail.com>
Message-ID: <CAKbxfG-LevpeEqmUU-VH5BhNE3=f_F5UcLMoYXGH48HK51tMeA@mail.gmail.com>

I believe I already answered this question when you asked on irc.  But
I'll answer it again so it can be logged.

To put it simply you are barking up the wrong tree trying to do
asynchronous things from a lua script.
Change your paradigm in your brain to think about it more like a GUI loop.

You only have 1 thread and you want to make the channel do something
by itself in the background but your script is already executing....


If you want this you need 2 threads.  There are several ways to get
this but you have to choose one.


1) Use ESL and an external application that can be running in your own
process so you can perform async operations.
2) Send the channel to the "park" app but spawn your lua script with
luarun via the dialplan variable.

<action application="eval" data="${luarun myscript.lua ${uuid}}"/>
<action application="park"/>

Then use the uuid argv[0] as the arg to a new session constructor that
will hook the channel to your object and you can tell the channel to
do things as long as you only execute them async
session:execute("foo::bar")
Remember if you are not running the session in its own thread this
will block and the whole idea is sort of messy but its possible to do
this.. if you script dies for instance your channel will not hangup.



3) Write your program in C where you have full access to threads
etc..... (THIS IS THE BEST CHOICE IMHO)
















On Tue, Oct 4, 2011 at 7:46 PM, Vu Quang Hoa <vutamhoan at gmail.com> wrote:
> Hi everyone,
> I've been trying ways to provide MOH in background of Lua script (searching
> and found few way) but none of them really?satisfied me. I think it
> may?implement in the same way of record_session function. So I need
> your?advises?if it is possible or not? It's a really good idea?
> Thank you!
> vutamhoan
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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 kheimerl at cs.berkeley.edu  Fri Oct  7 11:20:05 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Fri, 7 Oct 2011 00:20:05 -0700
Subject: [Freeswitch-dev] questions on the new mod_sms
In-Reply-To: <CAKbxfG_M1EMTuv0ms0g41cxGHM7LKryBci5a1Q_ti_wv6DbaBw@mail.gmail.com>
References: <BCE527BE963B4C78B9B3D9F3368211E5@gmail.com>
	<CAKbxfG9-BqaPXBtdikYpLXcVCBFk1dATP9BV1y=2yBnccmbwPg@mail.gmail.com>
	<CACyT-3m9Xf6L5kedwEmRpOryq=b0C_shLnickDQySw6aadVqxQ@mail.gmail.com>
	<8FB5B399218C4C04829DA071F604B665@gmail.com>
	<CAKzWOxXXZ8wZdCw3QvhBdJgu2qytz0b_VJUc7weUe2SpZ_UWRw@mail.gmail.com>
	<65819E89FA654014A1FA2754F365E62B@gmail.com>
	<CAKbxfG_M1EMTuv0ms0g41cxGHM7LKryBci5a1Q_ti_wv6DbaBw@mail.gmail.com>
Message-ID: <CACyT-3=ejFL0TOrWnrgxM2OsEXUzyJjdWdUCtmaas5e7DVZFUA@mail.gmail.com>

I'm loving this module so far. Does everything mine does, and more.
Glad to see this functionality supported in mainline.

I do have a question though. I'm having a hell of a time generating an
outgoing message. I tried the chat api function, but it seems to
generate a new walk down the chatplan (ending in a dp+ handler not
found error). I similarly can't generate a message from the CLI. Is
the "send" function going to provide this (and it's just not finished)
or am I missing something else? I'm building a little chat routing
chatplan, and I need to be able to redirect received messages to
another user.

I'll be happy to add to the wiki once this is figured out.

On Thu, Sep 29, 2011 at 8:45 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> message is a standard event so its the same api as always with extra
> method chat_execute to run the apps.
>
> message:chat_execute("reply", "you said: ${_body}");
>
>
>
>
> On Wed, Sep 28, 2011 at 11:59 PM, Seven Du <dujinfang at gmail.com> wrote:
>>
>> Seven,
>> Let me know if you have started a wiki page yet. I would like to help get
>> this documented. I need to write up a story on fs.org as well. I just
>> haven't had a chance to test drive this new module, but I will definitely be
>> doing so as soon as I can.
>> -MC
>>
>> See ?http://wiki.freeswitch.org/wiki/Mod_sms
>> It's incomplete. I haven't figured out how the "message" in lua works, e.g.
>> ?message:reply or message:execute(reply, blah) etc. ?Maybe it's haven't been
>> implemented or is there other ways to doing that?
>> Anyone have better idea feel free to update the page.
>> Thanks.
>> 7.
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> 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
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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 succer110 at tiscali.it  Fri Oct  7 16:13:16 2011
From: succer110 at tiscali.it (succer110 at tiscali.it)
Date: Fri, 7 Oct 2011 14:13:16 +0200 (CEST)
Subject: [Freeswitch-dev] lua executing application after hangup
Message-ID: <29621739.102741317989596455.JavaMail.defaultUser@defaultHost>

Hi, 	I'm writing some part of dialplan in lua and after bridge a have to execute an application.
I receive this: "Channel is hungup, aborting execution of application: antani-postdial". 
Of course i've tried all the kind of app interface flags.my dialstring contains right variable:2011-10-07 13:07:51.817103 [DEBUG] switch_event.c:1168 Parsing variable [continue_on_fail]=[true]2011-10-07 13:07:51.817103 [DEBUG] switch_event.c:1168 Parsing variable [hangup_after_bridge]=[false]
p.s.
in the docs i can find "SAF_ZOMBIE_EXEC" but it's not in my switch_types.h

what can i do to exec my postdial application in my lua?
dialstring="{ignore_early_media=true,originate_timeout=90,continue_on_fail=true,hangup_after_bridge=false}"..dialstringlocal sessionB = freeswitch.Session(dialstring);freeswitch.bridge(session, sessionB)session:execute("antani-postdial", i);


Thank you! =)



E' nata indoona: chiama, videochiama e messaggia Gratis. 
Scarica indoona per iPhone, Android e PC

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From anthony.minessale at gmail.com  Fri Oct  7 19:28:51 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 7 Oct 2011 10:28:51 -0500
Subject: [Freeswitch-dev] questions on the new mod_sms
In-Reply-To: <CACyT-3=ejFL0TOrWnrgxM2OsEXUzyJjdWdUCtmaas5e7DVZFUA@mail.gmail.com>
References: <BCE527BE963B4C78B9B3D9F3368211E5@gmail.com>
	<CAKbxfG9-BqaPXBtdikYpLXcVCBFk1dATP9BV1y=2yBnccmbwPg@mail.gmail.com>
	<CACyT-3m9Xf6L5kedwEmRpOryq=b0C_shLnickDQySw6aadVqxQ@mail.gmail.com>
	<8FB5B399218C4C04829DA071F604B665@gmail.com>
	<CAKzWOxXXZ8wZdCw3QvhBdJgu2qytz0b_VJUc7weUe2SpZ_UWRw@mail.gmail.com>
	<65819E89FA654014A1FA2754F365E62B@gmail.com>
	<CAKbxfG_M1EMTuv0ms0g41cxGHM7LKryBci5a1Q_ti_wv6DbaBw@mail.gmail.com>
	<CACyT-3=ejFL0TOrWnrgxM2OsEXUzyJjdWdUCtmaas5e7DVZFUA@mail.gmail.com>
Message-ID: <CAKbxfG-cEAgYdE9+2MCvOeJcYrN6cKOzApEYqgbtCbpA3H8Exg@mail.gmail.com>

I had a few bugs i guess.  Try again on latest it should work with chat api now.

Also here is a small esl perl script example:


require ESL;

my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon");

my $e = new ESL::ESLevent("custom", "SMS::SEND_MESSAGE");
$e->addHeader("to", "1004\@www.freeswitch.org");
$e->addHeader("from", "testing\@foobar.com");
$e->addHeader("sip_profile", "internal");
$e->addHeader("dest_proto", "sip");
$e->addBody(shift);

$con->sendEvent($e);



On Fri, Oct 7, 2011 at 2:20 AM, Kurtis Heimerl <kheimerl at cs.berkeley.edu> wrote:
> I'm loving this module so far. Does everything mine does, and more.
> Glad to see this functionality supported in mainline.
>
> I do have a question though. I'm having a hell of a time generating an
> outgoing message. I tried the chat api function, but it seems to
> generate a new walk down the chatplan (ending in a dp+ handler not
> found error). I similarly can't generate a message from the CLI. Is
> the "send" function going to provide this (and it's just not finished)
> or am I missing something else? I'm building a little chat routing
> chatplan, and I need to be able to redirect received messages to
> another user.
>
> I'll be happy to add to the wiki once this is figured out.
>
> On Thu, Sep 29, 2011 at 8:45 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> message is a standard event so its the same api as always with extra
>> method chat_execute to run the apps.
>>
>> message:chat_execute("reply", "you said: ${_body}");
>>
>>
>>
>>
>> On Wed, Sep 28, 2011 at 11:59 PM, Seven Du <dujinfang at gmail.com> wrote:
>>>
>>> Seven,
>>> Let me know if you have started a wiki page yet. I would like to help get
>>> this documented. I need to write up a story on fs.org as well. I just
>>> haven't had a chance to test drive this new module, but I will definitely be
>>> doing so as soon as I can.
>>> -MC
>>>
>>> See ?http://wiki.freeswitch.org/wiki/Mod_sms
>>> It's incomplete. I haven't figured out how the "message" in lua works, e.g.
>>> ?message:reply or message:execute(reply, blah) etc. ?Maybe it's haven't been
>>> implemented or is there other ways to doing that?
>>> Anyone have better idea feel free to update the page.
>>> Thanks.
>>> 7.
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> 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
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> 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
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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 Oct  7 21:14:12 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 7 Oct 2011 12:14:12 -0500
Subject: [Freeswitch-dev] lua executing application after hangup
In-Reply-To: <29621739.102741317989596455.JavaMail.defaultUser@defaultHost>
References: <29621739.102741317989596455.JavaMail.defaultUser@defaultHost>
Message-ID: <CAKbxfG9Ciwh2kgzxnvAdEy6fw=GuKG6Gk1tZL=_cmUTcYtHcag@mail.gmail.com>

you need to update to lastest git and execute the app set_zombie_exec
before the channel has hungup.



On Fri, Oct 7, 2011 at 7:13 AM, succer110 at tiscali.it
<succer110 at tiscali.it> wrote:
> Hi,
> I'm writing some part of dialplan in lua and after bridge a have to execute
> an application.
> I receive this:?"Channel is hungup, aborting execution of application:
> antani-postdial".
> Of course i've tried all the kind of app interface flags.
> my dialstring contains right variable:
> 2011-10-07 13:07:51.817103 [DEBUG] switch_event.c:1168 Parsing variable
> [continue_on_fail]=[true]
> 2011-10-07 13:07:51.817103 [DEBUG] switch_event.c:1168 Parsing variable
> [hangup_after_bridge]=[false]
> p.s.
> in the docs i can find "SAF_ZOMBIE_EXEC" but it's not in my switch_types.h
>
> what can i do to exec my postdial application in my lua?
>
> dialstring="{ignore_early_media=true,originate_timeout=90,continue_on_fail=true,hangup_after_bridge=false}"..dialstring
> local sessionB = freeswitch.Session(dialstring);
> freeswitch.bridge(session, sessionB)
> session:execute("antani-postdial", i);
>
>
> Thank you! =)
>
>
>
> E' nata indoona: chiama, videochiama e messaggia Gratis.
> Scarica indoona per iPhone, Android e PC
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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 kheimerl at cs.berkeley.edu  Fri Oct  7 22:04:01 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Fri, 7 Oct 2011 11:04:01 -0700
Subject: [Freeswitch-dev] questions on the new mod_sms
In-Reply-To: <CAKbxfG-cEAgYdE9+2MCvOeJcYrN6cKOzApEYqgbtCbpA3H8Exg@mail.gmail.com>
References: <BCE527BE963B4C78B9B3D9F3368211E5@gmail.com>
	<CAKbxfG9-BqaPXBtdikYpLXcVCBFk1dATP9BV1y=2yBnccmbwPg@mail.gmail.com>
	<CACyT-3m9Xf6L5kedwEmRpOryq=b0C_shLnickDQySw6aadVqxQ@mail.gmail.com>
	<8FB5B399218C4C04829DA071F604B665@gmail.com>
	<CAKzWOxXXZ8wZdCw3QvhBdJgu2qytz0b_VJUc7weUe2SpZ_UWRw@mail.gmail.com>
	<65819E89FA654014A1FA2754F365E62B@gmail.com>
	<CAKbxfG_M1EMTuv0ms0g41cxGHM7LKryBci5a1Q_ti_wv6DbaBw@mail.gmail.com>
	<CACyT-3=ejFL0TOrWnrgxM2OsEXUzyJjdWdUCtmaas5e7DVZFUA@mail.gmail.com>
	<CAKbxfG-cEAgYdE9+2MCvOeJcYrN6cKOzApEYqgbtCbpA3H8Exg@mail.gmail.com>
Message-ID: <CACyT-3mYXO+V-uAptPBsCsAJgVUg26n-De=Q+GNvkGRpzydaCQ@mail.gmail.com>

I think the build is broken:

cc1: warnings being treated as errors
src/switch_loadable_module.c: In function ?do_chat_send?:
src/switch_loadable_module.c:552: error: format not a string literal
and no format arguments
make[1]: *** [libfreeswitch_la-switch_loadable_module.lo] Error 1
make: *** [all] Error 2

I tried a fresh configure and am seeing the same issue, so I don't
think it's me being an idiot.

On Fri, Oct 7, 2011 at 8:28 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> I had a few bugs i guess. ?Try again on latest it should work with chat api now.
>
> Also here is a small esl perl script example:
>
>
> require ESL;
>
> my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon");
>
> my $e = new ESL::ESLevent("custom", "SMS::SEND_MESSAGE");
> $e->addHeader("to", "1004\@www.freeswitch.org");
> $e->addHeader("from", "testing\@foobar.com");
> $e->addHeader("sip_profile", "internal");
> $e->addHeader("dest_proto", "sip");
> $e->addBody(shift);
>
> $con->sendEvent($e);
>
>
>
> On Fri, Oct 7, 2011 at 2:20 AM, Kurtis Heimerl <kheimerl at cs.berkeley.edu> wrote:
>> I'm loving this module so far. Does everything mine does, and more.
>> Glad to see this functionality supported in mainline.
>>
>> I do have a question though. I'm having a hell of a time generating an
>> outgoing message. I tried the chat api function, but it seems to
>> generate a new walk down the chatplan (ending in a dp+ handler not
>> found error). I similarly can't generate a message from the CLI. Is
>> the "send" function going to provide this (and it's just not finished)
>> or am I missing something else? I'm building a little chat routing
>> chatplan, and I need to be able to redirect received messages to
>> another user.
>>
>> I'll be happy to add to the wiki once this is figured out.
>>
>> On Thu, Sep 29, 2011 at 8:45 AM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>>> message is a standard event so its the same api as always with extra
>>> method chat_execute to run the apps.
>>>
>>> message:chat_execute("reply", "you said: ${_body}");
>>>
>>>
>>>
>>>
>>> On Wed, Sep 28, 2011 at 11:59 PM, Seven Du <dujinfang at gmail.com> wrote:
>>>>
>>>> Seven,
>>>> Let me know if you have started a wiki page yet. I would like to help get
>>>> this documented. I need to write up a story on fs.org as well. I just
>>>> haven't had a chance to test drive this new module, but I will definitely be
>>>> doing so as soon as I can.
>>>> -MC
>>>>
>>>> See ?http://wiki.freeswitch.org/wiki/Mod_sms
>>>> It's incomplete. I haven't figured out how the "message" in lua works, e.g.
>>>> ?message:reply or message:execute(reply, blah) etc. ?Maybe it's haven't been
>>>> implemented or is there other ways to doing that?
>>>> Anyone have better idea feel free to update the page.
>>>> Thanks.
>>>> 7.
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> 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
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> 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
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> 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
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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 Oct  7 22:23:34 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 7 Oct 2011 13:23:34 -0500
Subject: [Freeswitch-dev] questions on the new mod_sms
In-Reply-To: <CACyT-3mYXO+V-uAptPBsCsAJgVUg26n-De=Q+GNvkGRpzydaCQ@mail.gmail.com>
References: <BCE527BE963B4C78B9B3D9F3368211E5@gmail.com>
	<CAKbxfG9-BqaPXBtdikYpLXcVCBFk1dATP9BV1y=2yBnccmbwPg@mail.gmail.com>
	<CACyT-3m9Xf6L5kedwEmRpOryq=b0C_shLnickDQySw6aadVqxQ@mail.gmail.com>
	<8FB5B399218C4C04829DA071F604B665@gmail.com>
	<CAKzWOxXXZ8wZdCw3QvhBdJgu2qytz0b_VJUc7weUe2SpZ_UWRw@mail.gmail.com>
	<65819E89FA654014A1FA2754F365E62B@gmail.com>
	<CAKbxfG_M1EMTuv0ms0g41cxGHM7LKryBci5a1Q_ti_wv6DbaBw@mail.gmail.com>
	<CACyT-3=ejFL0TOrWnrgxM2OsEXUzyJjdWdUCtmaas5e7DVZFUA@mail.gmail.com>
	<CAKbxfG-cEAgYdE9+2MCvOeJcYrN6cKOzApEYqgbtCbpA3H8Exg@mail.gmail.com>
	<CACyT-3mYXO+V-uAptPBsCsAJgVUg26n-De=Q+GNvkGRpzydaCQ@mail.gmail.com>
Message-ID: <CAKbxfG9TKTpUgi-kinXZ644DZ+fvS5qE2JovWRquwN+tYccTDw@mail.gmail.com>

hm.... my compiler didn't mind that, fixed .....


On Fri, Oct 7, 2011 at 1:04 PM, Kurtis Heimerl <kheimerl at cs.berkeley.edu> wrote:
> I think the build is broken:
>
> cc1: warnings being treated as errors
> src/switch_loadable_module.c: In function ?do_chat_send?:
> src/switch_loadable_module.c:552: error: format not a string literal
> and no format arguments
> make[1]: *** [libfreeswitch_la-switch_loadable_module.lo] Error 1
> make: *** [all] Error 2
>
> I tried a fresh configure and am seeing the same issue, so I don't
> think it's me being an idiot.
>
> On Fri, Oct 7, 2011 at 8:28 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> I had a few bugs i guess. ?Try again on latest it should work with chat api now.
>>
>> Also here is a small esl perl script example:
>>
>>
>> require ESL;
>>
>> my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon");
>>
>> my $e = new ESL::ESLevent("custom", "SMS::SEND_MESSAGE");
>> $e->addHeader("to", "1004\@www.freeswitch.org");
>> $e->addHeader("from", "testing\@foobar.com");
>> $e->addHeader("sip_profile", "internal");
>> $e->addHeader("dest_proto", "sip");
>> $e->addBody(shift);
>>
>> $con->sendEvent($e);
>>
>>
>>
>> On Fri, Oct 7, 2011 at 2:20 AM, Kurtis Heimerl <kheimerl at cs.berkeley.edu> wrote:
>>> I'm loving this module so far. Does everything mine does, and more.
>>> Glad to see this functionality supported in mainline.
>>>
>>> I do have a question though. I'm having a hell of a time generating an
>>> outgoing message. I tried the chat api function, but it seems to
>>> generate a new walk down the chatplan (ending in a dp+ handler not
>>> found error). I similarly can't generate a message from the CLI. Is
>>> the "send" function going to provide this (and it's just not finished)
>>> or am I missing something else? I'm building a little chat routing
>>> chatplan, and I need to be able to redirect received messages to
>>> another user.
>>>
>>> I'll be happy to add to the wiki once this is figured out.
>>>
>>> On Thu, Sep 29, 2011 at 8:45 AM, Anthony Minessale
>>> <anthony.minessale at gmail.com> wrote:
>>>> message is a standard event so its the same api as always with extra
>>>> method chat_execute to run the apps.
>>>>
>>>> message:chat_execute("reply", "you said: ${_body}");
>>>>
>>>>
>>>>
>>>>
>>>> On Wed, Sep 28, 2011 at 11:59 PM, Seven Du <dujinfang at gmail.com> wrote:
>>>>>
>>>>> Seven,
>>>>> Let me know if you have started a wiki page yet. I would like to help get
>>>>> this documented. I need to write up a story on fs.org as well. I just
>>>>> haven't had a chance to test drive this new module, but I will definitely be
>>>>> doing so as soon as I can.
>>>>> -MC
>>>>>
>>>>> See ?http://wiki.freeswitch.org/wiki/Mod_sms
>>>>> It's incomplete. I haven't figured out how the "message" in lua works, e.g.
>>>>> ?message:reply or message:execute(reply, blah) etc. ?Maybe it's haven't been
>>>>> implemented or is there other ways to doing that?
>>>>> Anyone have better idea feel free to update the page.
>>>>> Thanks.
>>>>> 7.
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> 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
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> 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
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> 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
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> 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
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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 kheimerl at cs.berkeley.edu  Fri Oct  7 22:41:59 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Fri, 7 Oct 2011 11:41:59 -0700
Subject: [Freeswitch-dev] questions on the new mod_sms
In-Reply-To: <CAKbxfG9TKTpUgi-kinXZ644DZ+fvS5qE2JovWRquwN+tYccTDw@mail.gmail.com>
References: <BCE527BE963B4C78B9B3D9F3368211E5@gmail.com>
	<CAKbxfG9-BqaPXBtdikYpLXcVCBFk1dATP9BV1y=2yBnccmbwPg@mail.gmail.com>
	<CACyT-3m9Xf6L5kedwEmRpOryq=b0C_shLnickDQySw6aadVqxQ@mail.gmail.com>
	<8FB5B399218C4C04829DA071F604B665@gmail.com>
	<CAKzWOxXXZ8wZdCw3QvhBdJgu2qytz0b_VJUc7weUe2SpZ_UWRw@mail.gmail.com>
	<65819E89FA654014A1FA2754F365E62B@gmail.com>
	<CAKbxfG_M1EMTuv0ms0g41cxGHM7LKryBci5a1Q_ti_wv6DbaBw@mail.gmail.com>
	<CACyT-3=ejFL0TOrWnrgxM2OsEXUzyJjdWdUCtmaas5e7DVZFUA@mail.gmail.com>
	<CAKbxfG-cEAgYdE9+2MCvOeJcYrN6cKOzApEYqgbtCbpA3H8Exg@mail.gmail.com>
	<CACyT-3mYXO+V-uAptPBsCsAJgVUg26n-De=Q+GNvkGRpzydaCQ@mail.gmail.com>
	<CAKbxfG9TKTpUgi-kinXZ644DZ+fvS5qE2JovWRquwN+tYccTDw@mail.gmail.com>
Message-ID: <CACyT-3nPwc++MzdnyHz-Jc4T2XFVii7j7M-fVma-L9C3ZTzeOA@mail.gmail.com>

Build worked, thanks! I'll see if chat works this afternoon.

On Fri, Oct 7, 2011 at 11:23 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> hm.... my compiler didn't mind that, fixed .....
>
>
> On Fri, Oct 7, 2011 at 1:04 PM, Kurtis Heimerl <kheimerl at cs.berkeley.edu> wrote:
>> I think the build is broken:
>>
>> cc1: warnings being treated as errors
>> src/switch_loadable_module.c: In function ?do_chat_send?:
>> src/switch_loadable_module.c:552: error: format not a string literal
>> and no format arguments
>> make[1]: *** [libfreeswitch_la-switch_loadable_module.lo] Error 1
>> make: *** [all] Error 2
>>
>> I tried a fresh configure and am seeing the same issue, so I don't
>> think it's me being an idiot.
>>
>> On Fri, Oct 7, 2011 at 8:28 AM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>>> I had a few bugs i guess. ?Try again on latest it should work with chat api now.
>>>
>>> Also here is a small esl perl script example:
>>>
>>>
>>> require ESL;
>>>
>>> my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon");
>>>
>>> my $e = new ESL::ESLevent("custom", "SMS::SEND_MESSAGE");
>>> $e->addHeader("to", "1004\@www.freeswitch.org");
>>> $e->addHeader("from", "testing\@foobar.com");
>>> $e->addHeader("sip_profile", "internal");
>>> $e->addHeader("dest_proto", "sip");
>>> $e->addBody(shift);
>>>
>>> $con->sendEvent($e);
>>>
>>>
>>>
>>> On Fri, Oct 7, 2011 at 2:20 AM, Kurtis Heimerl <kheimerl at cs.berkeley.edu> wrote:
>>>> I'm loving this module so far. Does everything mine does, and more.
>>>> Glad to see this functionality supported in mainline.
>>>>
>>>> I do have a question though. I'm having a hell of a time generating an
>>>> outgoing message. I tried the chat api function, but it seems to
>>>> generate a new walk down the chatplan (ending in a dp+ handler not
>>>> found error). I similarly can't generate a message from the CLI. Is
>>>> the "send" function going to provide this (and it's just not finished)
>>>> or am I missing something else? I'm building a little chat routing
>>>> chatplan, and I need to be able to redirect received messages to
>>>> another user.
>>>>
>>>> I'll be happy to add to the wiki once this is figured out.
>>>>
>>>> On Thu, Sep 29, 2011 at 8:45 AM, Anthony Minessale
>>>> <anthony.minessale at gmail.com> wrote:
>>>>> message is a standard event so its the same api as always with extra
>>>>> method chat_execute to run the apps.
>>>>>
>>>>> message:chat_execute("reply", "you said: ${_body}");
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Wed, Sep 28, 2011 at 11:59 PM, Seven Du <dujinfang at gmail.com> wrote:
>>>>>>
>>>>>> Seven,
>>>>>> Let me know if you have started a wiki page yet. I would like to help get
>>>>>> this documented. I need to write up a story on fs.org as well. I just
>>>>>> haven't had a chance to test drive this new module, but I will definitely be
>>>>>> doing so as soon as I can.
>>>>>> -MC
>>>>>>
>>>>>> See ?http://wiki.freeswitch.org/wiki/Mod_sms
>>>>>> It's incomplete. I haven't figured out how the "message" in lua works, e.g.
>>>>>> ?message:reply or message:execute(reply, blah) etc. ?Maybe it's haven't been
>>>>>> implemented or is there other ways to doing that?
>>>>>> Anyone have better idea feel free to update the page.
>>>>>> Thanks.
>>>>>> 7.
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> 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
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> 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
>>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> 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
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> 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
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> 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
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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 vutamhoan at gmail.com  Thu Oct  6 04:37:17 2011
From: vutamhoan at gmail.com (Vu Quang Hoa)
Date: Thu, 6 Oct 2011 07:37:17 +0700
Subject: [Freeswitch-dev] trouble on how to remove the media bug from
 the session after the sessions are closed
In-Reply-To: <94715a0a3f69399835a066ea4d8ea62c@gw.ind.rwth-aachen.de>
References: <94715a0a3f69399835a066ea4d8ea62c@gw.ind.rwth-aachen.de>
Message-ID: <CAJnMD2eXz--h=HZrGBgXi8Wz9h+ic-vGStcrBMUORaEYG8P-aQ@mail.gmail.com>

As I tested - FS automatically removes bug when call ends.
I'm trying to figure out if it does the same with allocated resources in
add_bug routine.

On Tue, Oct 4, 2011 at 7:03 PM, Han Xin <xin at ind.rwth-aachen.de> wrote:

> Hello everyone!
>
> I am new to Freeswitch and try to add media bugs such as digital signal
> processing to the read stream and write stream. But I have some trouble on
> how to remove the media bug from the session after the sessions are closed.
> I tried to look into other existed modules such as mod_soundtouch and
> mod_dptools to find a solution, the former mod_soundtouch put the function
> "switch_core_media_bug_remove()" in the beginning of application run
> function. The latter mod use a single app called "remove bugs" and put this
> app in the dial plan to remove the bugs.
>
> The question is where should I put this "switch_core_media_bug_remove()"
> function after the sessions are closed.
> Is there a way to automatically remove a media bug when a call is ended?
> Or it is only possible to use a single app to remove the media bugs as in
> the mod_dptools?
>
> Thank you all in advance!
>
> --
>
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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 Oct  7 04:54:52 2011
From: vutamhoan at gmail.com (Vu Quang Hoa)
Date: Fri, 7 Oct 2011 07:54:52 +0700
Subject: [Freeswitch-dev] MOH in async mode is possible?
In-Reply-To: <CAKbxfG-LevpeEqmUU-VH5BhNE3=f_F5UcLMoYXGH48HK51tMeA@mail.gmail.com>
References: <CAJnMD2cRRu8FC_W0Jt5hyPmourSxG=X+6a7R5eVga1V6mZZxbw@mail.gmail.com>
	<CAKbxfG-LevpeEqmUU-VH5BhNE3=f_F5UcLMoYXGH48HK51tMeA@mail.gmail.com>
Message-ID: <CAJnMD2f0+L0iNAsSnUzXWfGF2KgwQseFzGk52LBcrvvp3jQbYg@mail.gmail.com>

Thanks Anthony,

I sent this message before chat with you on irc. Of cause I tried to search
& search many time before asking you (I love Google) and I had already tried
ESL but it still blocks any LUA session call (like session:foo()).

Now I'm trying to implement MOH with async mode like record_session concept
(using mod_soundtouch sample code).

Hope that it will be ready within few days.

-vutamhoan

On Thu, Oct 6, 2011 at 10:08 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> I believe I already answered this question when you asked on irc.  But
> I'll answer it again so it can be logged.
>
> To put it simply you are barking up the wrong tree trying to do
> asynchronous things from a lua script.
> Change your paradigm in your brain to think about it more like a GUI loop.
>
> You only have 1 thread and you want to make the channel do something
> by itself in the background but your script is already executing....
>
>
> If you want this you need 2 threads.  There are several ways to get
> this but you have to choose one.
>
>
> 1) Use ESL and an external application that can be running in your own
> process so you can perform async operations.
> 2) Send the channel to the "park" app but spawn your lua script with
> luarun via the dialplan variable.
>
> <action application="eval" data="${luarun myscript.lua ${uuid}}"/>
> <action application="park"/>
>
> Then use the uuid argv[0] as the arg to a new session constructor that
> will hook the channel to your object and you can tell the channel to
> do things as long as you only execute them async
> session:execute("foo::bar")
> Remember if you are not running the session in its own thread this
> will block and the whole idea is sort of messy but its possible to do
> this.. if you script dies for instance your channel will not hangup.
>
>
>
> 3) Write your program in C where you have full access to threads
> etc..... (THIS IS THE BEST CHOICE IMHO)
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> On Tue, Oct 4, 2011 at 7:46 PM, Vu Quang Hoa <vutamhoan at gmail.com> wrote:
> > Hi everyone,
> > I've been trying ways to provide MOH in background of Lua script
> (searching
> > and found few way) but none of them really satisfied me. I think it
> > may implement in the same way of record_session function. So I need
> > your advises if it is possible or not? It's a really good idea?
> > Thank you!
> > vutamhoan
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > 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
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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 anthony.minessale at gmail.com  Mon Oct 10 19:17:36 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 10 Oct 2011 10:17:36 -0500
Subject: [Freeswitch-dev] trouble on how to remove the media bug from
 the session after the sessions are closed
In-Reply-To: <CAJnMD2eXz--h=HZrGBgXi8Wz9h+ic-vGStcrBMUORaEYG8P-aQ@mail.gmail.com>
References: <94715a0a3f69399835a066ea4d8ea62c@gw.ind.rwth-aachen.de>
	<CAJnMD2eXz--h=HZrGBgXi8Wz9h+ic-vGStcrBMUORaEYG8P-aQ@mail.gmail.com>
Message-ID: <CAKbxfG-hv-TnFpTq0n9kbGkSh1KJh-OdOpJSo6XhDfE-KRmtug@mail.gmail.com>

it calls the callback with type set to SWITCH_ABC_TYPE_CLOSE

On Wed, Oct 5, 2011 at 7:37 PM, Vu Quang Hoa <vutamhoan at gmail.com> wrote:
> As I tested - FS automatically removes bug when call ends.
> I'm trying to?figure?out if it does the same with allocated resources in
> add_bug routine.
>
> On Tue, Oct 4, 2011 at 7:03 PM, Han Xin <xin at ind.rwth-aachen.de> wrote:
>>
>> Hello everyone!
>>
>> I am new to Freeswitch and try to add media bugs such as digital signal
>> processing to the read stream and write stream. But I have some trouble on
>> how to remove the media bug from the session after the sessions are closed.
>> I tried to look into other existed modules such as mod_soundtouch and
>> mod_dptools to find a solution, the former mod_soundtouch put the function
>> "switch_core_media_bug_remove()" in the beginning of application run
>> function. The latter mod use a single app called "remove bugs" and put this
>> app in the dial plan to remove the bugs.
>>
>> The question is where should I put this "switch_core_media_bug_remove()"
>> function after the sessions are closed.
>> Is there a way to automatically remove a media bug when a call is ended?
>> Or it is only possible to use a single app to remove the media bugs as in
>> the mod_dptools?
>>
>> Thank you all in advance!
>>
>> --
>>
>>
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> 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
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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 msc at freeswitch.org  Wed Oct 12 21:05:05 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 12 Oct 2011 10:05:05 -0700
Subject: [Freeswitch-dev] FreeSWITCH Conference Call Right Now! - Phil
	Zimmermann
Message-ID: <CAKzWOxVekoCX+y961zkTkPD5VGFDHU-38C72OqYcW9QgcCZerA@mail.gmail.com>

Hey all,

It seems my first email to the list didn't make it. Just a reminder that
Philip Zimmermann of PGP fame is coming in to talk about VoIP security, ZRTP
and Zfone. Please join us now!

-Michael
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From jcherukuri_necc at yahoo.com  Thu Oct 13 00:50:48 2011
From: jcherukuri_necc at yahoo.com (Jyotshna Cherukuri)
Date: Wed, 12 Oct 2011 13:50:48 -0700 (PDT)
Subject: [Freeswitch-dev] Gracefully terminate ON HOLD party with
	"rtp-hold-timeout-sec" parameter
In-Reply-To: <1315515545.44633.YahooMailNeo@web122515.mail.ne1.yahoo.com>
References: <1315515545.44633.YahooMailNeo@web122515.mail.ne1.yahoo.com>
Message-ID: <1318452648.62651.YahooMailNeo@web122510.mail.ne1.yahoo.com>

Hi,

I was wondering if there is any way I could configure in the dialplan to gracefully terminate the ON HOLD party by playing a recording ("Please try the call again later as the other party is experiencing some network isseus")at the end of hold music rather than simply disconnecting the call as what is doing right now. Any help is greatly appreciated.



Thanks in advance
Regards

Jyotshna
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From lg at perfectitchina.com  Mon Oct 10 18:37:00 2011
From: lg at perfectitchina.com (lg)
Date: Mon, 10 Oct 2011 10:37:00 -0400
Subject: [Freeswitch-dev] FreeSWITCH dialstring/channel variables
Message-ID: <CA+B+5Z+pL7=LtE9gx78XnLNMbeGOCip=7MSJmfyzrR3ySKkz9w@mail.gmail.com>

One quick question here. Is it any dialstring to control dial function? Like
as,i would change the dial legs sequence as dial caller number first
and then when caller pickup the call, dial the called destination number (
as the same as "originate dial_string destination_number" function). Your
hint or short response would be much appreciated.


Thanks,

Grant
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From xin at ind.rwth-aachen.de  Mon Oct 10 20:03:33 2011
From: xin at ind.rwth-aachen.de (Han Xin)
Date: Mon, 10 Oct 2011 18:03:33 +0200
Subject: [Freeswitch-dev] trouble on how to remove the media bug from
 the session after the sessions are closed
In-Reply-To: <CAKbxfG-hv-TnFpTq0n9kbGkSh1KJh-OdOpJSo6XhDfE-KRmtug@mail.gmail.com>
Message-ID: <ac15452888a19e5b352295fc10eccf15@gw.ind.rwth-aachen.de>

 yes, I use the SWITCH_ABC_TYPE_CLOSE type and I've found the media bugs are automatically removed with the DEBUG log infomation. Now the problem is the session ending.

Detailed situation description: I added two media bugs to both local session and remote session of one call. When I ended the call, the media bugs are removed first, and then the session should be ended. But only the local session is ended correctly. 
When I press F3 and F4 to show current channels and calls, the remote channel and call info is till there. But when I try to dump uuid to get channel infomation with the cmd "uuid_dump remote_session_uuid", it says there is no such channel!

here is some related log info:

2011-10-10 17:36:41.428463 [ERR] mod_bwe.c:345 28b3a542-aecd-4b9b-9f55-3c2674a25b5c: Remove local variable store!
2011-10-10 17:36:41.428463 [DEBUG] switch_ivr_bridge.c:1332 sofia/internal/sip:1001 at 137.226.198.252 :5060 skip receive message [UNBRIDGE] (channel is hungup already)
2011-10-10 17:36:41.428463 [DEBUG] switch_ivr_bridge.c:1335 sofia/internal/1000 at asterisk skip receive message [UNBRIDGE] (channel is hungup already)
2011-10-10 17:36:41.428463 [DEBUG] switch_core_media_bug.c:480 Removing BUG from sofia/internal/sip:1001 at 137.226.198.252 :5060
2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/sip:1001 at 137.226.198.252 :5060) State HANGUP
2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:454 sofia/internal/sip:1001 at 137.226.198.252 :5060 Overriding SIP cause 480 with 200 from the other leg
2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:460 Channel sofia/internal/sip:1001 at 137.226.198.252 :5060 hanging up, cause: NORMAL_CLEARING
2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:503 Sending BYE to sofia/internal/sip:1001 at 137.226.198.252 :5060
2011-10-10 17:36:41.428463 [DEBUG] switch_core_session.c:2235 sofia/internal/1000 at asterisk skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/1000 at asterisk ) State EXECUTE going to sleep
2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1000 at asterisk ) Running State Change CS_HANGUP
2011-10-10 17:36:41.428463 [ERR] mod_bwe.c:345 713d5f0c-4341-47ec-b0bf-4b14bd6b4726: Remove local variable store!
2011-10-10 17:36:41.428463 [DEBUG] switch_core_media_bug.c:480 Removing BUG from sofia/internal/1000 at asterisk 
2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1000 at asterisk ) State HANGUP
2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:454 sofia/internal/1000 at asterisk Overriding SIP cause 480 with 200 from the other leg
2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:460 Channel sofia/internal/1000 at asterisk hanging up, cause: NORMAL_CLEARING
2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:47 sofia/internal/1000 at asterisk Standard HANGUP, cause: NORMAL_CLEARING
2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:47 sofia/internal/sip:1001 at 137.226.198.252 :5060 Standard HANGUP, cause: NORMAL_CLEARING
2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/sip:1001 at 137.226.198.252 :5060) State HANGUP going to sleep
2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/sip:1001 at 137.226.198.252 :5060) State Change CS_HANGUP -> CS_REPORTING
2011-10-10 17:36:41.428463 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/sip:1001 at 137.226.198.252 :5060 [BREAK]
2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1001 at 137.226.198.252 :5060) Running State Change CS_REPORTING
2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/sip:1001 at 137.226.198.252 :5060) State REPORTING
2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:79 sofia/internal/sip:1001 at 137.226.198.252 :5060 Standard REPORTING, cause: NORMAL_CLEARING
2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/sip:1001 at 137.226.198.252 :5060) State REPORTING going to sleep
2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/sip:1001 at 137.226.198.252 :5060) State Change CS_REPORTING -> CS_DESTROY
2011-10-10 17:36:41.428463 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/sip:1001 at 137.226.198.252 :5060 [BREAK]
2011-10-10 17:36:41.428463 [DEBUG] switch_core_session.c:1351 Session 8 (sofia/internal/sip:1001 at 137.226.198.252 :5060) Locked, Waiting on external entities
2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1000 at asterisk ) State HANGUP going to sleep
2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/1000 at asterisk ) State Change CS_HANGUP -> CS_REPORTING
2011-10-10 17:36:41.428463 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/1000 at asterisk [BREAK]
2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1000 at asterisk ) Running State Change CS_REPORTING
2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/1000 at asterisk ) State REPORTING
2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:79 sofia/internal/1000 at asterisk Standard REPORTING, cause: NORMAL_CLEARING
2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/1000 at asterisk ) State REPORTING going to sleep
2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/1000 at asterisk ) State Change CS_REPORTING -> CS_DESTROY
2011-10-10 17:36:41.428463 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/1000 at asterisk [BREAK]
2011-10-10 17:36:41.428463 [DEBUG] switch_core_session.c:1351 Session 7 (sofia/internal/1000 at asterisk ) Locked, Waiting on external entities
2011-10-10 17:36:41.428463 [NOTICE] switch_core_session.c:1369 Session 7 (sofia/internal/1000 at asterisk ) Ended
2011-10-10 17:36:41.428463 [NOTICE] switch_core_session.c:1371 Close Channel sofia/internal/1000 at asterisk [CS_DESTROY]
2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/1000 at asterisk ) Callstate Change HANGUP -> DOWN
2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/1000 at asterisk ) Running State Change CS_DESTROY
2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1000 at asterisk ) State DESTROY
2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:365 sofia/internal/1000 at asterisk SOFIA DESTROY
2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:86 sofia/internal/1000 at asterisk Standard DESTROY
2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1000 at asterisk ) State DESTROY going to sleep

Could anyone give me some hints why the remote session is not ended automatically?

 
----------------urspr?ngliche Nachricht-----------------
Von: "Anthony Minessale" anthony.minessale at gmail.com 
An: freeswitch-dev at lists.freeswitch.org 
Datum: Mon, 10 Oct 2011 10:17:36 -0500
-------------------------------------------------
 
 
> it calls the callback with type set to SWITCH_ABC_TYPE_CLOSE
> 
> On Wed, Oct 5, 2011 at 7:37 PM, Vu Quang Hoa vutamhoan at gmail.com wrote:
>> As I tested - FS automatically removes bug when call ends.
>> I'm trying to figure out if it does the same with allocated resources in
>> add_bug routine.
>>
>> On Tue, Oct 4, 2011 at 7:03 PM, Han Xin xin at ind.rwth-aachen.de wrote:
>>>
>>> Hello everyone!
>>>
>>> I am new to Freeswitch and try to add media bugs such as digital signal
>>> processing to the read stream and write stream. But I have some trouble on
>>> how to remove the media bug from the session after the sessions are 
>>> closed.
>>> I tried to look into other existed modules such as mod_soundtouch and
>>> mod_dptools to find a solution, the former mod_soundtouch put the 
>>> function
>>> "switch_core_media_bug_remove()" in the beginning of application 
>>> run
>>> function. The latter mod use a single app called "remove bugs" and put 
>>> this
>>> app in the dial plan to remove the bugs.
>>>
>>> The question is where should I put this 
>>> "switch_core_media_bug_remove()"
>>> function after the sessions are closed.
>>> Is there a way to automatically remove a media bug when a call is ended?
>>> Or it is only possible to use a single app to remove the media bugs as in
>>> the mod_dptools?
>>>
>>> Thank you all in advance!
>>>
>>> --
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> 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/freeswit
>>> ch-dev
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> 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
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> 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-d
> ev
> http://www.freeswitch.org
> 

-- 

 



From anthony.minessale at gmail.com  Thu Oct 13 02:06:41 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 12 Oct 2011 17:06:41 -0500
Subject: [Freeswitch-dev] trouble on how to remove the media bug from
 the session after the sessions are closed
In-Reply-To: <ac15452888a19e5b352295fc10eccf15@gw.ind.rwth-aachen.de>
References: <CAKbxfG-hv-TnFpTq0n9kbGkSh1KJh-OdOpJSo6XhDfE-KRmtug@mail.gmail.com>
	<ac15452888a19e5b352295fc10eccf15@gw.ind.rwth-aachen.de>
Message-ID: <CAKbxfG_FARW_78QRj3TBzQZzAwZ-nGhk7QyLHi3f+2c806-Xqg@mail.gmail.com>

looks like you have some custom mods maybe you used
switch_core_session_locate and did not rwunlock it?
are you using latest GIT ?


On Mon, Oct 10, 2011 at 11:03 AM, Han Xin <xin at ind.rwth-aachen.de> wrote:
> ?yes, I use the SWITCH_ABC_TYPE_CLOSE type and I've found the media bugs are automatically removed with the DEBUG log infomation. Now the problem is the session ending.
>
> Detailed situation description: I added two media bugs to both local session and remote session of one call. When I ended the call, the media bugs are removed first, and then the session should be ended. But only the local session is ended correctly.
> When I press F3 and F4 to show current channels and calls, the remote channel and call info is till there. But when I try to dump uuid to get channel infomation with the cmd "uuid_dump remote_session_uuid", it says there is no such channel!
>
> here is some related log info:
>
> 2011-10-10 17:36:41.428463 [ERR] mod_bwe.c:345 28b3a542-aecd-4b9b-9f55-3c2674a25b5c: Remove local variable store!
> 2011-10-10 17:36:41.428463 [DEBUG] switch_ivr_bridge.c:1332 sofia/internal/sip:1001 at 137.226.198.252 :5060 skip receive message [UNBRIDGE] (channel is hungup already)
> 2011-10-10 17:36:41.428463 [DEBUG] switch_ivr_bridge.c:1335 sofia/internal/1000 at asterisk skip receive message [UNBRIDGE] (channel is hungup already)
> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_media_bug.c:480 Removing BUG from sofia/internal/sip:1001 at 137.226.198.252 :5060
> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/sip:1001 at 137.226.198.252 :5060) State HANGUP
> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:454 sofia/internal/sip:1001 at 137.226.198.252 :5060 Overriding SIP cause 480 with 200 from the other leg
> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:460 Channel sofia/internal/sip:1001 at 137.226.198.252 :5060 hanging up, cause: NORMAL_CLEARING
> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:503 Sending BYE to sofia/internal/sip:1001 at 137.226.198.252 :5060
> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_session.c:2235 sofia/internal/1000 at asterisk skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/1000 at asterisk ) State EXECUTE going to sleep
> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1000 at asterisk ) Running State Change CS_HANGUP
> 2011-10-10 17:36:41.428463 [ERR] mod_bwe.c:345 713d5f0c-4341-47ec-b0bf-4b14bd6b4726: Remove local variable store!
> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_media_bug.c:480 Removing BUG from sofia/internal/1000 at asterisk
> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1000 at asterisk ) State HANGUP
> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:454 sofia/internal/1000 at asterisk Overriding SIP cause 480 with 200 from the other leg
> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:460 Channel sofia/internal/1000 at asterisk hanging up, cause: NORMAL_CLEARING
> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:47 sofia/internal/1000 at asterisk Standard HANGUP, cause: NORMAL_CLEARING
> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:47 sofia/internal/sip:1001 at 137.226.198.252 :5060 Standard HANGUP, cause: NORMAL_CLEARING
> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/sip:1001 at 137.226.198.252 :5060) State HANGUP going to sleep
> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/sip:1001 at 137.226.198.252 :5060) State Change CS_HANGUP -> CS_REPORTING
> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/sip:1001 at 137.226.198.252 :5060 [BREAK]
> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1001 at 137.226.198.252 :5060) Running State Change CS_REPORTING
> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/sip:1001 at 137.226.198.252 :5060) State REPORTING
> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:79 sofia/internal/sip:1001 at 137.226.198.252 :5060 Standard REPORTING, cause: NORMAL_CLEARING
> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/sip:1001 at 137.226.198.252 :5060) State REPORTING going to sleep
> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/sip:1001 at 137.226.198.252 :5060) State Change CS_REPORTING -> CS_DESTROY
> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/sip:1001 at 137.226.198.252 :5060 [BREAK]
> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_session.c:1351 Session 8 (sofia/internal/sip:1001 at 137.226.198.252 :5060) Locked, Waiting on external entities
> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1000 at asterisk ) State HANGUP going to sleep
> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/1000 at asterisk ) State Change CS_HANGUP -> CS_REPORTING
> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/1000 at asterisk [BREAK]
> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1000 at asterisk ) Running State Change CS_REPORTING
> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/1000 at asterisk ) State REPORTING
> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:79 sofia/internal/1000 at asterisk Standard REPORTING, cause: NORMAL_CLEARING
> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/1000 at asterisk ) State REPORTING going to sleep
> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/1000 at asterisk ) State Change CS_REPORTING -> CS_DESTROY
> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/1000 at asterisk [BREAK]
> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_session.c:1351 Session 7 (sofia/internal/1000 at asterisk ) Locked, Waiting on external entities
> 2011-10-10 17:36:41.428463 [NOTICE] switch_core_session.c:1369 Session 7 (sofia/internal/1000 at asterisk ) Ended
> 2011-10-10 17:36:41.428463 [NOTICE] switch_core_session.c:1371 Close Channel sofia/internal/1000 at asterisk [CS_DESTROY]
> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/1000 at asterisk ) Callstate Change HANGUP -> DOWN
> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/1000 at asterisk ) Running State Change CS_DESTROY
> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1000 at asterisk ) State DESTROY
> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:365 sofia/internal/1000 at asterisk SOFIA DESTROY
> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:86 sofia/internal/1000 at asterisk Standard DESTROY
> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1000 at asterisk ) State DESTROY going to sleep
>
> Could anyone give me some hints why the remote session is not ended automatically?
>
>
> ----------------urspr?ngliche Nachricht-----------------
> Von: "Anthony Minessale" anthony.minessale at gmail.com
> An: freeswitch-dev at lists.freeswitch.org
> Datum: Mon, 10 Oct 2011 10:17:36 -0500
> -------------------------------------------------
>
>
>> it calls the callback with type set to SWITCH_ABC_TYPE_CLOSE
>>
>> On Wed, Oct 5, 2011 at 7:37 PM, Vu Quang Hoa vutamhoan at gmail.com wrote:
>>> As I tested - FS automatically removes bug when call ends.
>>> I'm trying to figure out if it does the same with allocated resources in
>>> add_bug routine.
>>>
>>> On Tue, Oct 4, 2011 at 7:03 PM, Han Xin xin at ind.rwth-aachen.de wrote:
>>>>
>>>> Hello everyone!
>>>>
>>>> I am new to Freeswitch and try to add media bugs such as digital signal
>>>> processing to the read stream and write stream. But I have some trouble on
>>>> how to remove the media bug from the session after the sessions are
>>>> closed.
>>>> I tried to look into other existed modules such as mod_soundtouch and
>>>> mod_dptools to find a solution, the former mod_soundtouch put the
>>>> function
>>>> "switch_core_media_bug_remove()" in the beginning of application
>>>> run
>>>> function. The latter mod use a single app called "remove bugs" and put
>>>> this
>>>> app in the dial plan to remove the bugs.
>>>>
>>>> The question is where should I put this
>>>> "switch_core_media_bug_remove()"
>>>> function after the sessions are closed.
>>>> Is there a way to automatically remove a media bug when a call is ended?
>>>> Or it is only possible to use a single app to remove the media bugs as in
>>>> the mod_dptools?
>>>>
>>>> Thank you all in advance!
>>>>
>>>> --
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> 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/freeswit
>>>> ch-dev
>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> 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
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> 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-d
>> ev
>> http://www.freeswitch.org
>>
>
> --
>
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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 sunwood360 at gmail.com  Thu Oct 13 04:13:44 2011
From: sunwood360 at gmail.com (envelopes envelopes)
Date: Wed, 12 Oct 2011 17:13:44 -0700
Subject: [Freeswitch-dev] FreeSWITCH Conference Call Right Now! - Phil
	Zimmermann
In-Reply-To: <CAKzWOxVekoCX+y961zkTkPD5VGFDHU-38C72OqYcW9QgcCZerA@mail.gmail.com>
References: <CAKzWOxVekoCX+y961zkTkPD5VGFDHU-38C72OqYcW9QgcCZerA@mail.gmail.com>
Message-ID: <CAKq5mS7RwzNyN8O1UxXa4fXxOeEy7Tg-jMHHTtkwqchP+k-E7g@mail.gmail.com>

Missed the event, is there a recorded archive?
On Oct 12, 2011 10:06 AM, "Michael Collins" <msc at freeswitch.org> wrote:

> Hey all,
>
> It seems my first email to the list didn't make it. Just a reminder that
> Philip Zimmermann of PGP fame is coming in to talk about VoIP security, ZRTP
> and Zfone. Please join us now!
>
> -Michael
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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 xin at ind.rwth-aachen.de  Thu Oct 13 11:19:13 2011
From: xin at ind.rwth-aachen.de (Han Xin)
Date: Thu, 13 Oct 2011 09:19:13 +0200
Subject: [Freeswitch-dev] trouble on how to remove the media bug from
 the session after the sessions are closed
In-Reply-To: <CAKbxfG_FARW_78QRj3TBzQZzAwZ-nGhk7QyLHi3f+2c806-Xqg@mail.gmail.com>
Message-ID: <185efa8a4b35596b4e59ca33f58b19a0@gw.ind.rwth-aachen.de>

 yes I use a custom mod and the latest git on Oct 10 but I did not use the function "switch_core_session_locate".
 
----------------urspr?ngliche Nachricht-----------------
Von: "Anthony Minessale" anthony.minessale at gmail.com 
An: freeswitch-dev at lists.freeswitch.org 
Datum: Wed, 12 Oct 2011 17:06:41 -0500
-------------------------------------------------
 
 
> looks like you have some custom mods maybe you used
> switch_core_session_locate and did not rwunlock it?
> are you using latest GIT ?
> 
> 
> On Mon, Oct 10, 2011 at 11:03 AM, Han Xin xin at ind.rwth-aachen.de wrote:
>> yes, I use the SWITCH_ABC_TYPE_CLOSE type and I've found the media bugs are 
>> automatically removed with the DEBUG log infomation. Now the problem is the 
>> session ending.
>>
>> Detailed situation description: I added two media bugs to both local session 
>> and remote session of one call. When I ended the call, the media bugs are removed 
>> first, and then the session should be ended. But only the local session is ended 
>> correctly.
>> When I press F3 and F4 to show current channels and calls, the remote channel 
>> and call info is till there. But when I try to dump uuid to get channel infomation 
>> with the cmd "uuid_dump remote_session_uuid", it says there is no such 
>> channel!
>>
>> here is some related log info:
>>
>> 2011-10-10 17:36:41.428463 [ERR] mod_bwe.c:345 
>> 28b3a542-aecd-4b9b-9f55-3c2674a25b5c: Remove local variable store!
>> 2011-10-10 17:36:41.428463 [DEBUG] switch_ivr_bridge.c:1332 
>> sofia/internal/sip:1001 at 137.226.198.252 :5060 skip receive message 
>> [UNBRIDGE] (channel is hungup already)
>> 2011-10-10 17:36:41.428463 [DEBUG] switch_ivr_bridge.c:1335 
>> sofia/internal/1000 at asterisk skip receive message [UNBRIDGE] (channel is 
>> hungup already)
>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_media_bug.c:480 
>> Removing BUG from sofia/internal/sip:1001 at 137.226.198.252 :5060
>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:602 
>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State HANGUP
>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:454 
>> sofia/internal/sip:1001 at 137.226.198.252 :5060 Overriding SIP cause 480 
>> with 200 from the other leg
>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:460 Channel 
>> sofia/internal/sip:1001 at 137.226.198.252 :5060 hanging up, cause: 
>> NORMAL_CLEARING
>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:503 Sending BYE to 
>> sofia/internal/sip:1001 at 137.226.198.252 :5060
>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_session.c:2235 
>> sofia/internal/1000 at asterisk skip receive message 
>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:417 
>> (sofia/internal/1000 at asterisk ) State EXECUTE going to sleep
>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:362 
>> (sofia/internal/1000 at asterisk ) Running State Change CS_HANGUP
>> 2011-10-10 17:36:41.428463 [ERR] mod_bwe.c:345 
>> 713d5f0c-4341-47ec-b0bf-4b14bd6b4726: Remove local variable store!
>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_media_bug.c:480 
>> Removing BUG from sofia/internal/1000 at asterisk 
>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:602 
>> (sofia/internal/1000 at asterisk ) State HANGUP
>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:454 
>> sofia/internal/1000 at asterisk Overriding SIP cause 480 with 200 from the 
>> other leg
>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:460 Channel 
>> sofia/internal/1000 at asterisk hanging up, cause: NORMAL_CLEARING
>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:47 
>> sofia/internal/1000 at asterisk Standard HANGUP, cause: NORMAL_CLEARING
>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:47 
>> sofia/internal/sip:1001 at 137.226.198.252 :5060 Standard HANGUP, cause: 
>> NORMAL_CLEARING
>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:602 
>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State HANGUP going to 
>> sleep
>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:393 
>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State Change CS_HANGUP 
>> -> CS_REPORTING
>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_session.c:1177 Send 
>> signal sofia/internal/sip:1001 at 137.226.198.252 :5060 [BREAK]
>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:362 
>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) Running State Change 
>> CS_REPORTING
>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:662 
>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State REPORTING
>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:79 
>> sofia/internal/sip:1001 at 137.226.198.252 :5060 Standard REPORTING, 
>> cause: NORMAL_CLEARING
>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:662 
>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State REPORTING going 
>> to sleep
>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:387 
>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State Change 
>> CS_REPORTING -> CS_DESTROY
>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_session.c:1177 Send 
>> signal sofia/internal/sip:1001 at 137.226.198.252 :5060 [BREAK]
>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_session.c:1351 Session 
>> 8 (sofia/internal/sip:1001 at 137.226.198.252 :5060) Locked, Waiting on 
>> external entities
>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:602 
>> (sofia/internal/1000 at asterisk ) State HANGUP going to sleep
>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:393 
>> (sofia/internal/1000 at asterisk ) State Change CS_HANGUP -> CS_REPORTING
>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_session.c:1177 Send 
>> signal sofia/internal/1000 at asterisk [BREAK]
>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:362 
>> (sofia/internal/1000 at asterisk ) Running State Change CS_REPORTING
>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:662 
>> (sofia/internal/1000 at asterisk ) State REPORTING
>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:79 
>> sofia/internal/1000 at asterisk Standard REPORTING, cause: 
>> NORMAL_CLEARING
>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:662 
>> (sofia/internal/1000 at asterisk ) State REPORTING going to sleep
>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:387 
>> (sofia/internal/1000 at asterisk ) State Change CS_REPORTING -> CS_DESTROY
>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_session.c:1177 Send 
>> signal sofia/internal/1000 at asterisk [BREAK]
>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_session.c:1351 Session 
>> 7 (sofia/internal/1000 at asterisk ) Locked, Waiting on external entities
>> 2011-10-10 17:36:41.428463 [NOTICE] switch_core_session.c:1369 
>> Session 7 (sofia/internal/1000 at asterisk ) Ended
>> 2011-10-10 17:36:41.428463 [NOTICE] switch_core_session.c:1371 Close 
>> Channel sofia/internal/1000 at asterisk [CS_DESTROY]
>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:491 
>> (sofia/internal/1000 at asterisk ) Callstate Change HANGUP -> DOWN
>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:494 
>> (sofia/internal/1000 at asterisk ) Running State Change CS_DESTROY
>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:504 
>> (sofia/internal/1000 at asterisk ) State DESTROY
>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:365 
>> sofia/internal/1000 at asterisk SOFIA DESTROY
>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:86 
>> sofia/internal/1000 at asterisk Standard DESTROY
>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:504 
>> (sofia/internal/1000 at asterisk ) State DESTROY going to sleep
>>
>> Could anyone give me some hints why the remote session is not ended 
>> automatically?
>>
>>
>> ----------------urspr?ngliche Nachricht-----------------
>> Von: "Anthony Minessale" anthony.minessale at gmail.com 
>> An: freeswitch-dev at lists.freeswitch.org 
>> Datum: Mon, 10 Oct 2011 10:17:36 -0500
>> -------------------------------------------------
>>
>>
>>> it calls the callback with type set to SWITCH_ABC_TYPE_CLOSE
>>>
>>> On Wed, Oct 5, 2011 at 7:37 PM, Vu Quang Hoa vutamhoan at gmail.com wrote:
>>>> As I tested - FS automatically removes bug when call ends.
>>>> I'm trying to figure out if it does the same with allocated resources 
>>>> in
>>>> add_bug routine.
>>>>
>>>> On Tue, Oct 4, 2011 at 7:03 PM, Han Xin xin at ind.rwth-aachen.de wrote:
>>>>>
>>>>> Hello everyone!
>>>>>
>>>>> I am new to Freeswitch and try to add media bugs such as digital 
>>>>> signal
>>>>> processing to the read stream and write stream. But I have some 
>>>>> trouble on
>>>>> how to remove the media bug from the session after the sessions are
>>>>> closed.
>>>>> I tried to look into other existed modules such as mod_soundtouch 
>>>>> and
>>>>> mod_dptools to find a solution, the former mod_soundtouch put 
>>>>> the
>>>>> function
>>>>> "switch_core_media_bug_remove()" in the beginning of 
>>>>> application
>>>>> run
>>>>> function. The latter mod use a single app called "remove bugs" and 
>>>>> put
>>>>> this
>>>>> app in the dial plan to remove the bugs.
>>>>>
>>>>> The question is where should I put this
>>>>> "switch_core_media_bug_remove()"
>>>>> function after the sessions are closed.
>>>>> Is there a way to automatically remove a media bug when a call is 
>>>>> ended?
>>>>> Or it is only possible to use a single app to remove the media bugs as 
>>>>> in
>>>>> the mod_dptools?
>>>>>
>>>>> Thank you all in advance!
>>>>>
>>>>> --
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> 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/free
>>>>> swit
>>>>> ch-dev
>>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> 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/freesw
>>>> itch
>>>> -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
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> 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/freeswit
>>> ch-d
>>> ev
>>> http://www.freeswitch.org
>>>
>>
>> --
>>
>>
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> 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
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> 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-d
> ev
> http://www.freeswitch.org
> 

-- 

 



From gvmello at yx.cl  Thu Oct 13 20:27:26 2011
From: gvmello at yx.cl (Glaucio Vaz de Mello)
Date: Thu, 13 Oct 2011 13:27:26 -0300
Subject: [Freeswitch-dev] FreeTDM delay reconnecting E1/T1
Message-ID: <CAPLGU-71UcZxBD=M1CccVO0xYwNsRqH5Kk0KETyxZdmmHFtDvQ@mail.gmail.com>

Hi,

   I have a question for FreeTDM experts.
   When I have a T1/E1 ISDN-PRI link connected through Sangoma card, I can
check signalling and physical status using this command:
> ftdm sangoma_isdn show_spans wp1
span:wp1 physical:OK signalling:UP

   Unplugging E1/T1 cable some alarms are raised, but show_spans command
still returns "OK" and "UP".
   After some time (more than one minute) I get the disconnected status:
> ftdm sangoma_isdn show_spans wp1
span:wp1 physical:ALARMED signalling:DOWN

   Using sangoma wanpipemon command I can get disconnected status
immediately:
# wanpipemon -i w1g1 -c Ta
Loss of Signal:    ON

   My question is why FreeTDM takes this big delay to detect a disconnected
cable? Is there a way to reduce this time?
   The same happens when I reconnect the cable, FreeTMD takes a minute or
more to detect it.

thanks,
Glaucio
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From anthony.minessale at gmail.com  Fri Oct 14 00:55:35 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 13 Oct 2011 15:55:35 -0500
Subject: [Freeswitch-dev] trouble on how to remove the media bug from
 the session after the sessions are closed
In-Reply-To: <185efa8a4b35596b4e59ca33f58b19a0@gw.ind.rwth-aachen.de>
References: <CAKbxfG_FARW_78QRj3TBzQZzAwZ-nGhk7QyLHi3f+2c806-Xqg@mail.gmail.com>
	<185efa8a4b35596b4e59ca33f58b19a0@gw.ind.rwth-aachen.de>
Message-ID: <CAKbxfG98ojfsAdsN-9PC4_Ya9o4956pQdAOKt6VY76o-TLhAFw@mail.gmail.com>

how about switch_ivr_originate() ?

On Thu, Oct 13, 2011 at 2:19 AM, Han Xin <xin at ind.rwth-aachen.de> wrote:
> ?yes I use a custom mod and the latest git on Oct 10 but I did not use the function "switch_core_session_locate".
>
> ----------------urspr?ngliche Nachricht-----------------
> Von: "Anthony Minessale" anthony.minessale at gmail.com
> An: freeswitch-dev at lists.freeswitch.org
> Datum: Wed, 12 Oct 2011 17:06:41 -0500
> -------------------------------------------------
>
>
>> looks like you have some custom mods maybe you used
>> switch_core_session_locate and did not rwunlock it?
>> are you using latest GIT ?
>>
>>
>> On Mon, Oct 10, 2011 at 11:03 AM, Han Xin xin at ind.rwth-aachen.de wrote:
>>> yes, I use the SWITCH_ABC_TYPE_CLOSE type and I've found the media bugs are
>>> automatically removed with the DEBUG log infomation. Now the problem is the
>>> session ending.
>>>
>>> Detailed situation description: I added two media bugs to both local session
>>> and remote session of one call. When I ended the call, the media bugs are removed
>>> first, and then the session should be ended. But only the local session is ended
>>> correctly.
>>> When I press F3 and F4 to show current channels and calls, the remote channel
>>> and call info is till there. But when I try to dump uuid to get channel infomation
>>> with the cmd "uuid_dump remote_session_uuid", it says there is no such
>>> channel!
>>>
>>> here is some related log info:
>>>
>>> 2011-10-10 17:36:41.428463 [ERR] mod_bwe.c:345
>>> 28b3a542-aecd-4b9b-9f55-3c2674a25b5c: Remove local variable store!
>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_ivr_bridge.c:1332
>>> sofia/internal/sip:1001 at 137.226.198.252 :5060 skip receive message
>>> [UNBRIDGE] (channel is hungup already)
>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_ivr_bridge.c:1335
>>> sofia/internal/1000 at asterisk skip receive message [UNBRIDGE] (channel is
>>> hungup already)
>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_media_bug.c:480
>>> Removing BUG from sofia/internal/sip:1001 at 137.226.198.252 :5060
>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:602
>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State HANGUP
>>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:454
>>> sofia/internal/sip:1001 at 137.226.198.252 :5060 Overriding SIP cause 480
>>> with 200 from the other leg
>>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:460 Channel
>>> sofia/internal/sip:1001 at 137.226.198.252 :5060 hanging up, cause:
>>> NORMAL_CLEARING
>>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:503 Sending BYE to
>>> sofia/internal/sip:1001 at 137.226.198.252 :5060
>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_session.c:2235
>>> sofia/internal/1000 at asterisk skip receive message
>>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:417
>>> (sofia/internal/1000 at asterisk ) State EXECUTE going to sleep
>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:362
>>> (sofia/internal/1000 at asterisk ) Running State Change CS_HANGUP
>>> 2011-10-10 17:36:41.428463 [ERR] mod_bwe.c:345
>>> 713d5f0c-4341-47ec-b0bf-4b14bd6b4726: Remove local variable store!
>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_media_bug.c:480
>>> Removing BUG from sofia/internal/1000 at asterisk
>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:602
>>> (sofia/internal/1000 at asterisk ) State HANGUP
>>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:454
>>> sofia/internal/1000 at asterisk Overriding SIP cause 480 with 200 from the
>>> other leg
>>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:460 Channel
>>> sofia/internal/1000 at asterisk hanging up, cause: NORMAL_CLEARING
>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:47
>>> sofia/internal/1000 at asterisk Standard HANGUP, cause: NORMAL_CLEARING
>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:47
>>> sofia/internal/sip:1001 at 137.226.198.252 :5060 Standard HANGUP, cause:
>>> NORMAL_CLEARING
>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:602
>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State HANGUP going to
>>> sleep
>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:393
>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State Change CS_HANGUP
>>> -> CS_REPORTING
>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_session.c:1177 Send
>>> signal sofia/internal/sip:1001 at 137.226.198.252 :5060 [BREAK]
>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:362
>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) Running State Change
>>> CS_REPORTING
>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:662
>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State REPORTING
>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:79
>>> sofia/internal/sip:1001 at 137.226.198.252 :5060 Standard REPORTING,
>>> cause: NORMAL_CLEARING
>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:662
>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State REPORTING going
>>> to sleep
>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:387
>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State Change
>>> CS_REPORTING -> CS_DESTROY
>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_session.c:1177 Send
>>> signal sofia/internal/sip:1001 at 137.226.198.252 :5060 [BREAK]
>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_session.c:1351 Session
>>> 8 (sofia/internal/sip:1001 at 137.226.198.252 :5060) Locked, Waiting on
>>> external entities
>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:602
>>> (sofia/internal/1000 at asterisk ) State HANGUP going to sleep
>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:393
>>> (sofia/internal/1000 at asterisk ) State Change CS_HANGUP -> CS_REPORTING
>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_session.c:1177 Send
>>> signal sofia/internal/1000 at asterisk [BREAK]
>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:362
>>> (sofia/internal/1000 at asterisk ) Running State Change CS_REPORTING
>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:662
>>> (sofia/internal/1000 at asterisk ) State REPORTING
>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:79
>>> sofia/internal/1000 at asterisk Standard REPORTING, cause:
>>> NORMAL_CLEARING
>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:662
>>> (sofia/internal/1000 at asterisk ) State REPORTING going to sleep
>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:387
>>> (sofia/internal/1000 at asterisk ) State Change CS_REPORTING -> CS_DESTROY
>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_session.c:1177 Send
>>> signal sofia/internal/1000 at asterisk [BREAK]
>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_session.c:1351 Session
>>> 7 (sofia/internal/1000 at asterisk ) Locked, Waiting on external entities
>>> 2011-10-10 17:36:41.428463 [NOTICE] switch_core_session.c:1369
>>> Session 7 (sofia/internal/1000 at asterisk ) Ended
>>> 2011-10-10 17:36:41.428463 [NOTICE] switch_core_session.c:1371 Close
>>> Channel sofia/internal/1000 at asterisk [CS_DESTROY]
>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:491
>>> (sofia/internal/1000 at asterisk ) Callstate Change HANGUP -> DOWN
>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:494
>>> (sofia/internal/1000 at asterisk ) Running State Change CS_DESTROY
>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:504
>>> (sofia/internal/1000 at asterisk ) State DESTROY
>>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:365
>>> sofia/internal/1000 at asterisk SOFIA DESTROY
>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:86
>>> sofia/internal/1000 at asterisk Standard DESTROY
>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_state_machine.c:504
>>> (sofia/internal/1000 at asterisk ) State DESTROY going to sleep
>>>
>>> Could anyone give me some hints why the remote session is not ended
>>> automatically?
>>>
>>>
>>> ----------------urspr?ngliche Nachricht-----------------
>>> Von: "Anthony Minessale" anthony.minessale at gmail.com
>>> An: freeswitch-dev at lists.freeswitch.org
>>> Datum: Mon, 10 Oct 2011 10:17:36 -0500
>>> -------------------------------------------------
>>>
>>>
>>>> it calls the callback with type set to SWITCH_ABC_TYPE_CLOSE
>>>>
>>>> On Wed, Oct 5, 2011 at 7:37 PM, Vu Quang Hoa vutamhoan at gmail.com wrote:
>>>>> As I tested - FS automatically removes bug when call ends.
>>>>> I'm trying to figure out if it does the same with allocated resources
>>>>> in
>>>>> add_bug routine.
>>>>>
>>>>> On Tue, Oct 4, 2011 at 7:03 PM, Han Xin xin at ind.rwth-aachen.de wrote:
>>>>>>
>>>>>> Hello everyone!
>>>>>>
>>>>>> I am new to Freeswitch and try to add media bugs such as digital
>>>>>> signal
>>>>>> processing to the read stream and write stream. But I have some
>>>>>> trouble on
>>>>>> how to remove the media bug from the session after the sessions are
>>>>>> closed.
>>>>>> I tried to look into other existed modules such as mod_soundtouch
>>>>>> and
>>>>>> mod_dptools to find a solution, the former mod_soundtouch put
>>>>>> the
>>>>>> function
>>>>>> "switch_core_media_bug_remove()" in the beginning of
>>>>>> application
>>>>>> run
>>>>>> function. The latter mod use a single app called "remove bugs" and
>>>>>> put
>>>>>> this
>>>>>> app in the dial plan to remove the bugs.
>>>>>>
>>>>>> The question is where should I put this
>>>>>> "switch_core_media_bug_remove()"
>>>>>> function after the sessions are closed.
>>>>>> Is there a way to automatically remove a media bug when a call is
>>>>>> ended?
>>>>>> Or it is only possible to use a single app to remove the media bugs as
>>>>>> in
>>>>>> the mod_dptools?
>>>>>>
>>>>>> Thank you all in advance!
>>>>>>
>>>>>> --
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> 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/free
>>>>>> swit
>>>>>> ch-dev
>>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> 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/freesw
>>>>> itch
>>>>> -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
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> 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/freeswit
>>>> ch-d
>>>> ev
>>>> http://www.freeswitch.org
>>>>
>>>
>>> --
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> 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
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> 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-d
>> ev
>> http://www.freeswitch.org
>>
>
> --
>
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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 moises.silva at gmail.com  Sat Oct 15 20:32:48 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Sat, 15 Oct 2011 12:32:48 -0400
Subject: [Freeswitch-dev] FreeTDM delay reconnecting E1/T1
In-Reply-To: <CAPLGU-71UcZxBD=M1CccVO0xYwNsRqH5Kk0KETyxZdmmHFtDvQ@mail.gmail.com>
References: <CAPLGU-71UcZxBD=M1CccVO0xYwNsRqH5Kk0KETyxZdmmHFtDvQ@mail.gmail.com>
Message-ID: <CAA4nhyCs5rN54kUAz27RVCRY=fGGkshTHE96wtSt8oDFuF0bCg@mail.gmail.com>

On Thu, Oct 13, 2011 at 12:27 PM, Glaucio Vaz de Mello <gvmello at yx.cl>wrote:

>    My question is why FreeTDM takes this big delay to detect a disconnected
> cable? Is there a way to reduce this time?
>    The same happens when I reconnect the cable, FreeTMD takes a minute or
> more to detect it.
>

It should not be delayed. Did you try using "ftdm dump x x"? what does the
physical status says in that case? If you can contact me on IRC and give me
ssh I can take a look at your problem.

*Moises Silva
**Software Engineer, Development Manager***

msilva 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 x128

f.   +1 905 474 9223



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From xin at ind.rwth-aachen.de  Mon Oct 17 11:38:34 2011
From: xin at ind.rwth-aachen.de (Han Xin)
Date: Mon, 17 Oct 2011 09:38:34 +0200
Subject: [Freeswitch-dev] trouble on how to remove the media bug from
 the session after the sessions are closed
In-Reply-To: <CAKbxfG98ojfsAdsN-9PC4_Ya9o4956pQdAOKt6VY76o-TLhAFw@mail.gmail.com>
Message-ID: <6393481560ec47e5dbad4efaad442de3@gw.ind.rwth-aachen.de>

 no, no use of ivr_originate function. 
I constructed a helper struct which include the remote session, could this cause the session releasing problem?
 
----------------urspr?ngliche Nachricht-----------------
Von: "Anthony Minessale" anthony.minessale at gmail.com 
An: freeswitch-dev at lists.freeswitch.org 
Datum: Thu, 13 Oct 2011 15:55:35 -0500
-------------------------------------------------
 
 
> how about switch_ivr_originate() ?
> 
> On Thu, Oct 13, 2011 at 2:19 AM, Han Xin xin at ind.rwth-aachen.de wrote:
>> yes I use a custom mod and the latest git on Oct 10 but I did not use the function 
>> "switch_core_session_locate".
>>
>> ----------------urspr?ngliche Nachricht-----------------
>> Von: "Anthony Minessale" anthony.minessale at gmail.com 
>> An: freeswitch-dev at lists.freeswitch.org 
>> Datum: Wed, 12 Oct 2011 17:06:41 -0500
>> -------------------------------------------------
>>
>>
>>> looks like you have some custom mods maybe you used
>>> switch_core_session_locate and did not rwunlock it?
>>> are you using latest GIT ?
>>>
>>>
>>> On Mon, Oct 10, 2011 at 11:03 AM, Han Xin xin at ind.rwth-aachen.de wrote:
>>>> yes, I use the SWITCH_ABC_TYPE_CLOSE type and I've found the media 
>>>> bugs are
>>>> automatically removed with the DEBUG log infomation. Now the problem 
>>>> is the
>>>> session ending.
>>>>
>>>> Detailed situation description: I added two media bugs to both local 
>>>> session
>>>> and remote session of one call. When I ended the call, the media bugs are 
>>>> removed
>>>> first, and then the session should be ended. But only the local session 
>>>> is ended
>>>> correctly.
>>>> When I press F3 and F4 to show current channels and calls, the remote 
>>>> channel
>>>> and call info is till there. But when I try to dump uuid to get channel 
>>>> infomation
>>>> with the cmd "uuid_dump remote_session_uuid", it says there is no 
>>>> such
>>>> channel!
>>>>
>>>> here is some related log info:
>>>>
>>>> 2011-10-10 17:36:41.428463 [ERR] mod_bwe.c:345
>>>> 28b3a542-aecd-4b9b-9f55-3c2674a25b5c: Remove local variable 
>>>> store!
>>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_ivr_bridge.c:1332
>>>> sofia/internal/sip:1001 at 137.226.198.252 :5060 skip receive 
>>>> message
>>>> [UNBRIDGE] (channel is hungup already)
>>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_ivr_bridge.c:1335
>>>> sofia/internal/1000 at asterisk skip receive message [UNBRIDGE] 
>>>> (channel is
>>>> hungup already)
>>>> 2011-10-10 17:36:41.428463 [DEBUG] 
>>>> switch_core_media_bug.c:480
>>>> Removing BUG from sofia/internal/sip:1001 at 137.226.198.252 
>>>> :5060
>>>> 2011-10-10 17:36:41.428463 [DEBUG] 
>>>> switch_core_state_machine.c:602
>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State HANGUP
>>>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:454
>>>> sofia/internal/sip:1001 at 137.226.198.252 :5060 Overriding SIP 
>>>> cause 480
>>>> with 200 from the other leg
>>>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:460 Channel
>>>> sofia/internal/sip:1001 at 137.226.198.252 :5060 hanging up, 
>>>> cause:
>>>> NORMAL_CLEARING
>>>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:503 Sending BYE 
>>>> to
>>>> sofia/internal/sip:1001 at 137.226.198.252 :5060
>>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_session.c:2235
>>>> sofia/internal/1000 at asterisk skip receive message
>>>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>>>> 2011-10-10 17:36:41.428463 [DEBUG] 
>>>> switch_core_state_machine.c:417
>>>> (sofia/internal/1000 at asterisk ) State EXECUTE going to sleep
>>>> 2011-10-10 17:36:41.428463 [DEBUG] 
>>>> switch_core_state_machine.c:362
>>>> (sofia/internal/1000 at asterisk ) Running State Change CS_HANGUP
>>>> 2011-10-10 17:36:41.428463 [ERR] mod_bwe.c:345
>>>> 713d5f0c-4341-47ec-b0bf-4b14bd6b4726: Remove local variable 
>>>> store!
>>>> 2011-10-10 17:36:41.428463 [DEBUG] 
>>>> switch_core_media_bug.c:480
>>>> Removing BUG from sofia/internal/1000 at asterisk 
>>>> 2011-10-10 17:36:41.428463 [DEBUG] 
>>>> switch_core_state_machine.c:602
>>>> (sofia/internal/1000 at asterisk ) State HANGUP
>>>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:454
>>>> sofia/internal/1000 at asterisk Overriding SIP cause 480 with 200 
>>>> from the
>>>> other leg
>>>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:460 Channel
>>>> sofia/internal/1000 at asterisk hanging up, cause: 
>>>> NORMAL_CLEARING
>>>> 2011-10-10 17:36:41.428463 [DEBUG] 
>>>> switch_core_state_machine.c:47
>>>> sofia/internal/1000 at asterisk Standard HANGUP, cause: 
>>>> NORMAL_CLEARING
>>>> 2011-10-10 17:36:41.428463 [DEBUG] 
>>>> switch_core_state_machine.c:47
>>>> sofia/internal/sip:1001 at 137.226.198.252 :5060 Standard HANGUP, 
>>>> cause:
>>>> NORMAL_CLEARING
>>>> 2011-10-10 17:36:41.428463 [DEBUG] 
>>>> switch_core_state_machine.c:602
>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State HANGUP 
>>>> going to
>>>> sleep
>>>> 2011-10-10 17:36:41.428463 [DEBUG] 
>>>> switch_core_state_machine.c:393
>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State Change 
>>>> CS_HANGUP
>>>> -> CS_REPORTING
>>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_session.c:1177 
>>>> Send
>>>> signal sofia/internal/sip:1001 at 137.226.198.252 :5060 [BREAK]
>>>> 2011-10-10 17:36:41.428463 [DEBUG] 
>>>> switch_core_state_machine.c:362
>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) Running State 
>>>> Change
>>>> CS_REPORTING
>>>> 2011-10-10 17:36:41.428463 [DEBUG] 
>>>> switch_core_state_machine.c:662
>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State 
>>>> REPORTING
>>>> 2011-10-10 17:36:41.428463 [DEBUG] 
>>>> switch_core_state_machine.c:79
>>>> sofia/internal/sip:1001 at 137.226.198.252 :5060 Standard 
>>>> REPORTING,
>>>> cause: NORMAL_CLEARING
>>>> 2011-10-10 17:36:41.428463 [DEBUG] 
>>>> switch_core_state_machine.c:662
>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State 
>>>> REPORTING going
>>>> to sleep
>>>> 2011-10-10 17:36:41.428463 [DEBUG] 
>>>> switch_core_state_machine.c:387
>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State Change
>>>> CS_REPORTING -> CS_DESTROY
>>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_session.c:1177 
>>>> Send
>>>> signal sofia/internal/sip:1001 at 137.226.198.252 :5060 [BREAK]
>>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_session.c:1351 
>>>> Session
>>>> 8 (sofia/internal/sip:1001 at 137.226.198.252 :5060) Locked, 
>>>> Waiting on
>>>> external entities
>>>> 2011-10-10 17:36:41.428463 [DEBUG] 
>>>> switch_core_state_machine.c:602
>>>> (sofia/internal/1000 at asterisk ) State HANGUP going to sleep
>>>> 2011-10-10 17:36:41.428463 [DEBUG] 
>>>> switch_core_state_machine.c:393
>>>> (sofia/internal/1000 at asterisk ) State Change CS_HANGUP -> 
>>>> CS_REPORTING
>>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_session.c:1177 
>>>> Send
>>>> signal sofia/internal/1000 at asterisk [BREAK]
>>>> 2011-10-10 17:36:41.428463 [DEBUG] 
>>>> switch_core_state_machine.c:362
>>>> (sofia/internal/1000 at asterisk ) Running State Change 
>>>> CS_REPORTING
>>>> 2011-10-10 17:36:41.428463 [DEBUG] 
>>>> switch_core_state_machine.c:662
>>>> (sofia/internal/1000 at asterisk ) State REPORTING
>>>> 2011-10-10 17:36:41.428463 [DEBUG] 
>>>> switch_core_state_machine.c:79
>>>> sofia/internal/1000 at asterisk Standard REPORTING, cause:
>>>> NORMAL_CLEARING
>>>> 2011-10-10 17:36:41.428463 [DEBUG] 
>>>> switch_core_state_machine.c:662
>>>> (sofia/internal/1000 at asterisk ) State REPORTING going to sleep
>>>> 2011-10-10 17:36:41.428463 [DEBUG] 
>>>> switch_core_state_machine.c:387
>>>> (sofia/internal/1000 at asterisk ) State Change CS_REPORTING -> 
>>>> CS_DESTROY
>>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_session.c:1177 
>>>> Send
>>>> signal sofia/internal/1000 at asterisk [BREAK]
>>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_session.c:1351 
>>>> Session
>>>> 7 (sofia/internal/1000 at asterisk ) Locked, Waiting on external 
>>>> entities
>>>> 2011-10-10 17:36:41.428463 [NOTICE] 
>>>> switch_core_session.c:1369
>>>> Session 7 (sofia/internal/1000 at asterisk ) Ended
>>>> 2011-10-10 17:36:41.428463 [NOTICE] switch_core_session.c:1371 
>>>> Close
>>>> Channel sofia/internal/1000 at asterisk [CS_DESTROY]
>>>> 2011-10-10 17:36:41.428463 [DEBUG] 
>>>> switch_core_state_machine.c:491
>>>> (sofia/internal/1000 at asterisk ) Callstate Change HANGUP -> DOWN
>>>> 2011-10-10 17:36:41.428463 [DEBUG] 
>>>> switch_core_state_machine.c:494
>>>> (sofia/internal/1000 at asterisk ) Running State Change CS_DESTROY
>>>> 2011-10-10 17:36:41.428463 [DEBUG] 
>>>> switch_core_state_machine.c:504
>>>> (sofia/internal/1000 at asterisk ) State DESTROY
>>>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:365
>>>> sofia/internal/1000 at asterisk SOFIA DESTROY
>>>> 2011-10-10 17:36:41.428463 [DEBUG] 
>>>> switch_core_state_machine.c:86
>>>> sofia/internal/1000 at asterisk Standard DESTROY
>>>> 2011-10-10 17:36:41.428463 [DEBUG] 
>>>> switch_core_state_machine.c:504
>>>> (sofia/internal/1000 at asterisk ) State DESTROY going to sleep
>>>>
>>>> Could anyone give me some hints why the remote session is not ended
>>>> automatically?
>>>>
>>>>
>>>> ----------------urspr?ngliche Nachricht-----------------
>>>> Von: "Anthony Minessale" anthony.minessale at gmail.com 
>>>> An: freeswitch-dev at lists.freeswitch.org 
>>>> Datum: Mon, 10 Oct 2011 10:17:36 -0500
>>>> -------------------------------------------------
>>>>
>>>>
>>>>> it calls the callback with type set to SWITCH_ABC_TYPE_CLOSE
>>>>>
>>>>> On Wed, Oct 5, 2011 at 7:37 PM, Vu Quang Hoa vutamhoan at gmail.com 
>>>>> wrote:
>>>>>> As I tested - FS automatically removes bug when call ends.
>>>>>> I'm trying to figure out if it does the same with allocated 
>>>>>> resources
>>>>>> in
>>>>>> add_bug routine.
>>>>>>
>>>>>> On Tue, Oct 4, 2011 at 7:03 PM, Han Xin xin at ind.rwth-aachen.de 
>>>>>> wrote:
>>>>>>>
>>>>>>> Hello everyone!
>>>>>>>
>>>>>>> I am new to Freeswitch and try to add media bugs such as 
>>>>>>> digital
>>>>>>> signal
>>>>>>> processing to the read stream and write stream. But I have 
>>>>>>> some
>>>>>>> trouble on
>>>>>>> how to remove the media bug from the session after the 
>>>>>>> sessions are
>>>>>>> closed.
>>>>>>> I tried to look into other existed modules such as 
>>>>>>> mod_soundtouch
>>>>>>> and
>>>>>>> mod_dptools to find a solution, the former mod_soundtouch 
>>>>>>> put
>>>>>>> the
>>>>>>> function
>>>>>>> "switch_core_media_bug_remove()" in the beginning of
>>>>>>> application
>>>>>>> run
>>>>>>> function. The latter mod use a single app called "remove 
>>>>>>> bugs" and
>>>>>>> put
>>>>>>> this
>>>>>>> app in the dial plan to remove the bugs.
>>>>>>>
>>>>>>> The question is where should I put this
>>>>>>> "switch_core_media_bug_remove()"
>>>>>>> function after the sessions are closed.
>>>>>>> Is there a way to automatically remove a media bug when a call 
>>>>>>> is
>>>>>>> ended?
>>>>>>> Or it is only possible to use a single app to remove the media 
>>>>>>> bugs as
>>>>>>> in
>>>>>>> the mod_dptools?
>>>>>>>
>>>>>>> Thank you all in advance!
>>>>>>>
>>>>>>> --
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>
>>>>>>> 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/
>>>>>>> free
>>>>>>> swit
>>>>>>> ch-dev
>>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> FreeSWITCH-dev mailing list
>>>>>> FreeSWITCH-dev at lists.freeswitch.org 
>>>>>>
>>>>>> 
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-de
>>>>>> v
>>>>>>
>>>>>>
>>>>>> 
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/fr
>>>>>> eesw
>>>>>> itch
>>>>>> -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
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> 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/free
>>>>> swit
>>>>> ch-d
>>>>> ev
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>> --
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> 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/freesw
>>>> itch
>>>> -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
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> 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/freeswit
>>> ch-d
>>> ev
>>> http://www.freeswitch.org
>>>
>>
>> --
>>
>>
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> 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
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> 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-d
> ev
> http://www.freeswitch.org
> 

-- 

 



From anthony.minessale at gmail.com  Mon Oct 17 18:39:26 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 17 Oct 2011 09:39:26 -0500
Subject: [Freeswitch-dev] trouble on how to remove the media bug from
 the session after the sessions are closed
In-Reply-To: <6393481560ec47e5dbad4efaad442de3@gw.ind.rwth-aachen.de>
References: <CAKbxfG98ojfsAdsN-9PC4_Ya9o4956pQdAOKt6VY76o-TLhAFw@mail.gmail.com>
	<6393481560ec47e5dbad4efaad442de3@gw.ind.rwth-aachen.de>
Message-ID: <CAKbxfG9sRo3fVNAR+_+BoSv6rYsEhv7OKba8QVs8Yvsv7oR4_Q@mail.gmail.com>

I have no more guesses.  You need to review your custom code and find
out where you read locked your session.


On Mon, Oct 17, 2011 at 2:38 AM, Han Xin <xin at ind.rwth-aachen.de> wrote:
> ?no, no use of ivr_originate function.
> I constructed a helper struct which include the remote session, could this cause the session releasing problem?
>
> ----------------urspr?ngliche Nachricht-----------------
> Von: "Anthony Minessale" anthony.minessale at gmail.com
> An: freeswitch-dev at lists.freeswitch.org
> Datum: Thu, 13 Oct 2011 15:55:35 -0500
> -------------------------------------------------
>
>
>> how about switch_ivr_originate() ?
>>
>> On Thu, Oct 13, 2011 at 2:19 AM, Han Xin xin at ind.rwth-aachen.de wrote:
>>> yes I use a custom mod and the latest git on Oct 10 but I did not use the function
>>> "switch_core_session_locate".
>>>
>>> ----------------urspr?ngliche Nachricht-----------------
>>> Von: "Anthony Minessale" anthony.minessale at gmail.com
>>> An: freeswitch-dev at lists.freeswitch.org
>>> Datum: Wed, 12 Oct 2011 17:06:41 -0500
>>> -------------------------------------------------
>>>
>>>
>>>> looks like you have some custom mods maybe you used
>>>> switch_core_session_locate and did not rwunlock it?
>>>> are you using latest GIT ?
>>>>
>>>>
>>>> On Mon, Oct 10, 2011 at 11:03 AM, Han Xin xin at ind.rwth-aachen.de wrote:
>>>>> yes, I use the SWITCH_ABC_TYPE_CLOSE type and I've found the media
>>>>> bugs are
>>>>> automatically removed with the DEBUG log infomation. Now the problem
>>>>> is the
>>>>> session ending.
>>>>>
>>>>> Detailed situation description: I added two media bugs to both local
>>>>> session
>>>>> and remote session of one call. When I ended the call, the media bugs are
>>>>> removed
>>>>> first, and then the session should be ended. But only the local session
>>>>> is ended
>>>>> correctly.
>>>>> When I press F3 and F4 to show current channels and calls, the remote
>>>>> channel
>>>>> and call info is till there. But when I try to dump uuid to get channel
>>>>> infomation
>>>>> with the cmd "uuid_dump remote_session_uuid", it says there is no
>>>>> such
>>>>> channel!
>>>>>
>>>>> here is some related log info:
>>>>>
>>>>> 2011-10-10 17:36:41.428463 [ERR] mod_bwe.c:345
>>>>> 28b3a542-aecd-4b9b-9f55-3c2674a25b5c: Remove local variable
>>>>> store!
>>>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_ivr_bridge.c:1332
>>>>> sofia/internal/sip:1001 at 137.226.198.252 :5060 skip receive
>>>>> message
>>>>> [UNBRIDGE] (channel is hungup already)
>>>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_ivr_bridge.c:1335
>>>>> sofia/internal/1000 at asterisk skip receive message [UNBRIDGE]
>>>>> (channel is
>>>>> hungup already)
>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>> switch_core_media_bug.c:480
>>>>> Removing BUG from sofia/internal/sip:1001 at 137.226.198.252
>>>>> :5060
>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>> switch_core_state_machine.c:602
>>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State HANGUP
>>>>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:454
>>>>> sofia/internal/sip:1001 at 137.226.198.252 :5060 Overriding SIP
>>>>> cause 480
>>>>> with 200 from the other leg
>>>>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:460 Channel
>>>>> sofia/internal/sip:1001 at 137.226.198.252 :5060 hanging up,
>>>>> cause:
>>>>> NORMAL_CLEARING
>>>>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:503 Sending BYE
>>>>> to
>>>>> sofia/internal/sip:1001 at 137.226.198.252 :5060
>>>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_session.c:2235
>>>>> sofia/internal/1000 at asterisk skip receive message
>>>>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>> switch_core_state_machine.c:417
>>>>> (sofia/internal/1000 at asterisk ) State EXECUTE going to sleep
>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>> switch_core_state_machine.c:362
>>>>> (sofia/internal/1000 at asterisk ) Running State Change CS_HANGUP
>>>>> 2011-10-10 17:36:41.428463 [ERR] mod_bwe.c:345
>>>>> 713d5f0c-4341-47ec-b0bf-4b14bd6b4726: Remove local variable
>>>>> store!
>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>> switch_core_media_bug.c:480
>>>>> Removing BUG from sofia/internal/1000 at asterisk
>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>> switch_core_state_machine.c:602
>>>>> (sofia/internal/1000 at asterisk ) State HANGUP
>>>>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:454
>>>>> sofia/internal/1000 at asterisk Overriding SIP cause 480 with 200
>>>>> from the
>>>>> other leg
>>>>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:460 Channel
>>>>> sofia/internal/1000 at asterisk hanging up, cause:
>>>>> NORMAL_CLEARING
>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>> switch_core_state_machine.c:47
>>>>> sofia/internal/1000 at asterisk Standard HANGUP, cause:
>>>>> NORMAL_CLEARING
>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>> switch_core_state_machine.c:47
>>>>> sofia/internal/sip:1001 at 137.226.198.252 :5060 Standard HANGUP,
>>>>> cause:
>>>>> NORMAL_CLEARING
>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>> switch_core_state_machine.c:602
>>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State HANGUP
>>>>> going to
>>>>> sleep
>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>> switch_core_state_machine.c:393
>>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State Change
>>>>> CS_HANGUP
>>>>> -> CS_REPORTING
>>>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_session.c:1177
>>>>> Send
>>>>> signal sofia/internal/sip:1001 at 137.226.198.252 :5060 [BREAK]
>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>> switch_core_state_machine.c:362
>>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) Running State
>>>>> Change
>>>>> CS_REPORTING
>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>> switch_core_state_machine.c:662
>>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State
>>>>> REPORTING
>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>> switch_core_state_machine.c:79
>>>>> sofia/internal/sip:1001 at 137.226.198.252 :5060 Standard
>>>>> REPORTING,
>>>>> cause: NORMAL_CLEARING
>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>> switch_core_state_machine.c:662
>>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State
>>>>> REPORTING going
>>>>> to sleep
>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>> switch_core_state_machine.c:387
>>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State Change
>>>>> CS_REPORTING -> CS_DESTROY
>>>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_session.c:1177
>>>>> Send
>>>>> signal sofia/internal/sip:1001 at 137.226.198.252 :5060 [BREAK]
>>>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_session.c:1351
>>>>> Session
>>>>> 8 (sofia/internal/sip:1001 at 137.226.198.252 :5060) Locked,
>>>>> Waiting on
>>>>> external entities
>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>> switch_core_state_machine.c:602
>>>>> (sofia/internal/1000 at asterisk ) State HANGUP going to sleep
>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>> switch_core_state_machine.c:393
>>>>> (sofia/internal/1000 at asterisk ) State Change CS_HANGUP ->
>>>>> CS_REPORTING
>>>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_session.c:1177
>>>>> Send
>>>>> signal sofia/internal/1000 at asterisk [BREAK]
>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>> switch_core_state_machine.c:362
>>>>> (sofia/internal/1000 at asterisk ) Running State Change
>>>>> CS_REPORTING
>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>> switch_core_state_machine.c:662
>>>>> (sofia/internal/1000 at asterisk ) State REPORTING
>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>> switch_core_state_machine.c:79
>>>>> sofia/internal/1000 at asterisk Standard REPORTING, cause:
>>>>> NORMAL_CLEARING
>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>> switch_core_state_machine.c:662
>>>>> (sofia/internal/1000 at asterisk ) State REPORTING going to sleep
>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>> switch_core_state_machine.c:387
>>>>> (sofia/internal/1000 at asterisk ) State Change CS_REPORTING ->
>>>>> CS_DESTROY
>>>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_session.c:1177
>>>>> Send
>>>>> signal sofia/internal/1000 at asterisk [BREAK]
>>>>> 2011-10-10 17:36:41.428463 [DEBUG] switch_core_session.c:1351
>>>>> Session
>>>>> 7 (sofia/internal/1000 at asterisk ) Locked, Waiting on external
>>>>> entities
>>>>> 2011-10-10 17:36:41.428463 [NOTICE]
>>>>> switch_core_session.c:1369
>>>>> Session 7 (sofia/internal/1000 at asterisk ) Ended
>>>>> 2011-10-10 17:36:41.428463 [NOTICE] switch_core_session.c:1371
>>>>> Close
>>>>> Channel sofia/internal/1000 at asterisk [CS_DESTROY]
>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>> switch_core_state_machine.c:491
>>>>> (sofia/internal/1000 at asterisk ) Callstate Change HANGUP -> DOWN
>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>> switch_core_state_machine.c:494
>>>>> (sofia/internal/1000 at asterisk ) Running State Change CS_DESTROY
>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>> switch_core_state_machine.c:504
>>>>> (sofia/internal/1000 at asterisk ) State DESTROY
>>>>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:365
>>>>> sofia/internal/1000 at asterisk SOFIA DESTROY
>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>> switch_core_state_machine.c:86
>>>>> sofia/internal/1000 at asterisk Standard DESTROY
>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>> switch_core_state_machine.c:504
>>>>> (sofia/internal/1000 at asterisk ) State DESTROY going to sleep
>>>>>
>>>>> Could anyone give me some hints why the remote session is not ended
>>>>> automatically?
>>>>>
>>>>>
>>>>> ----------------urspr?ngliche Nachricht-----------------
>>>>> Von: "Anthony Minessale" anthony.minessale at gmail.com
>>>>> An: freeswitch-dev at lists.freeswitch.org
>>>>> Datum: Mon, 10 Oct 2011 10:17:36 -0500
>>>>> -------------------------------------------------
>>>>>
>>>>>
>>>>>> it calls the callback with type set to SWITCH_ABC_TYPE_CLOSE
>>>>>>
>>>>>> On Wed, Oct 5, 2011 at 7:37 PM, Vu Quang Hoa vutamhoan at gmail.com
>>>>>> wrote:
>>>>>>> As I tested - FS automatically removes bug when call ends.
>>>>>>> I'm trying to figure out if it does the same with allocated
>>>>>>> resources
>>>>>>> in
>>>>>>> add_bug routine.
>>>>>>>
>>>>>>> On Tue, Oct 4, 2011 at 7:03 PM, Han Xin xin at ind.rwth-aachen.de
>>>>>>> wrote:
>>>>>>>>
>>>>>>>> Hello everyone!
>>>>>>>>
>>>>>>>> I am new to Freeswitch and try to add media bugs such as
>>>>>>>> digital
>>>>>>>> signal
>>>>>>>> processing to the read stream and write stream. But I have
>>>>>>>> some
>>>>>>>> trouble on
>>>>>>>> how to remove the media bug from the session after the
>>>>>>>> sessions are
>>>>>>>> closed.
>>>>>>>> I tried to look into other existed modules such as
>>>>>>>> mod_soundtouch
>>>>>>>> and
>>>>>>>> mod_dptools to find a solution, the former mod_soundtouch
>>>>>>>> put
>>>>>>>> the
>>>>>>>> function
>>>>>>>> "switch_core_media_bug_remove()" in the beginning of
>>>>>>>> application
>>>>>>>> run
>>>>>>>> function. The latter mod use a single app called "remove
>>>>>>>> bugs" and
>>>>>>>> put
>>>>>>>> this
>>>>>>>> app in the dial plan to remove the bugs.
>>>>>>>>
>>>>>>>> The question is where should I put this
>>>>>>>> "switch_core_media_bug_remove()"
>>>>>>>> function after the sessions are closed.
>>>>>>>> Is there a way to automatically remove a media bug when a call
>>>>>>>> is
>>>>>>>> ended?
>>>>>>>> Or it is only possible to use a single app to remove the media
>>>>>>>> bugs as
>>>>>>>> in
>>>>>>>> the mod_dptools?
>>>>>>>>
>>>>>>>> Thank you all in advance!
>>>>>>>>
>>>>>>>> --
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>
>>>>>>>> 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/
>>>>>>>> free
>>>>>>>> swit
>>>>>>>> ch-dev
>>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>
>>>>>>> FreeSWITCH-dev mailing list
>>>>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-de
>>>>>>> v
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/fr
>>>>>>> eesw
>>>>>>> itch
>>>>>>> -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
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> 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/free
>>>>>> swit
>>>>>> ch-d
>>>>>> ev
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>> --
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> 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/freesw
>>>>> itch
>>>>> -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
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> 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/freeswit
>>>> ch-d
>>>> ev
>>>> http://www.freeswitch.org
>>>>
>>>
>>> --
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> 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
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> 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-d
>> ev
>> http://www.freeswitch.org
>>
>
> --
>
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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 gvmello at yx.cl  Mon Oct 17 18:50:07 2011
From: gvmello at yx.cl (Glaucio Vaz de Mello)
Date: Mon, 17 Oct 2011 11:50:07 -0300
Subject: [Freeswitch-dev] FreeTDM delay reconnecting E1/T1
In-Reply-To: <CAA4nhyCs5rN54kUAz27RVCRY=fGGkshTHE96wtSt8oDFuF0bCg@mail.gmail.com>
References: <CAPLGU-71UcZxBD=M1CccVO0xYwNsRqH5Kk0KETyxZdmmHFtDvQ@mail.gmail.com>
	<CAA4nhyCs5rN54kUAz27RVCRY=fGGkshTHE96wtSt8oDFuF0bCg@mail.gmail.com>
Message-ID: <CAPLGU-4xAjjgGWXtZHa+ukJ9vLZ0QtNz-knwmfoPc6gmngV6Aw@mail.gmail.com>

Hi,

Thanks for the help.
Yes, I tried ftdm dump and got the same results. This command was issued
after unplugging the cable:

freeswitch at internal> ftdm dump wp1 1
span_id: 1
chan_id: 1
physical_span_id: 1
physical_chan_id: 1
physical_status: ok
signaling_status: UP
type: B
state: DOWN
last_state: DOWN
txgain: 0.00
rxgain: 0.00
cid_date:
cid_name:
cid_num:
ani:
aniII:
dnis:
rdnis:
cause: NONE
session: (none)

-- No state history --




On Sat, Oct 15, 2011 at 1:32 PM, Moises Silva <moises.silva at gmail.com>wrote:

> On Thu, Oct 13, 2011 at 12:27 PM, Glaucio Vaz de Mello <gvmello at yx.cl>wrote:
>
>>    My question is why FreeTDM takes this big delay to detect a
>> disconnected cable? Is there a way to reduce this time?
>>    The same happens when I reconnect the cable, FreeTMD takes a minute or
>> more to detect it.
>>
>
> It should not be delayed. Did you try using "ftdm dump x x"? what does the
> physical status says in that case? If you can contact me on IRC and give me
> ssh I can take a look at your problem.
>
> *Moises Silva
> **Software Engineer, Development Manager***
>
> msilva 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 x128
>
> f.   +1 905 474 9223
>
>
>
> **<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email+signatures>
>
> Products<http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
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>  | Twitter<http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>`|
> | YouTube<http://www.youtube.com/sangomatechnologies?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>
>                 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>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
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> http://www.freeswitch.org
>
>
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From moises.silva at gmail.com  Mon Oct 17 20:16:51 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Mon, 17 Oct 2011 12:16:51 -0400
Subject: [Freeswitch-dev] FreeTDM delay reconnecting E1/T1
In-Reply-To: <CAPLGU-4xAjjgGWXtZHa+ukJ9vLZ0QtNz-knwmfoPc6gmngV6Aw@mail.gmail.com>
References: <CAPLGU-71UcZxBD=M1CccVO0xYwNsRqH5Kk0KETyxZdmmHFtDvQ@mail.gmail.com>
	<CAA4nhyCs5rN54kUAz27RVCRY=fGGkshTHE96wtSt8oDFuF0bCg@mail.gmail.com>
	<CAPLGU-4xAjjgGWXtZHa+ukJ9vLZ0QtNz-knwmfoPc6gmngV6Aw@mail.gmail.com>
Message-ID: <CAA4nhyCOBei_aEmtDHqeFAYiSztNpYmncv993ZgwF744GGHYEg@mail.gmail.com>

On Mon, Oct 17, 2011 at 10:50 AM, Glaucio Vaz de Mello <gvmello at yx.cl>wrote:

> Hi,
>
> Thanks for the help.
> Yes, I tried ftdm dump and got the same results. This command was issued
> after unplugging the cable:
>
> freeswitch at internal> ftdm dump wp1 1
> span_id: 1
> chan_id: 1
> physical_span_id: 1
> physical_chan_id: 1
> physical_status: ok
> signaling_status: UP
> type: B
> state: DOWN
> last_state: DOWN
> txgain: 0.00
> rxgain: 0.00
> cid_date:
> cid_name:
> cid_num:
> ani:
> aniII:
> dnis:
> rdnis:
> cause: NONE
> session: (none)
>
> -- No state history --
>
>
>
That 's odd. Are you using latest git HEAD? if you are using latest git
HEAD, ping me on IRC and I can take a quick look. If is not solved within 10
mins we'll have to open a JIRA ticket.

*Moises Silva
**Software Engineer, Development Manager***

msilva 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 x128

f.   +1 905 474 9223



**<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email+signatures>

Products<http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
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 | Twitter<http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>`|
| YouTube<http://www.youtube.com/sangomatechnologies?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>

                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>
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From moises.silva at gmail.com  Mon Oct 17 20:22:58 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Mon, 17 Oct 2011 12:22:58 -0400
Subject: [Freeswitch-dev] MTU setting and application buffer size
In-Reply-To: <CAAqVCyTaPsWz2tmShrpEgwFmY27nFaw0AOHDq3Mmn932QTQ4aQ@mail.gmail.com>
References: <CAAqVCyTaPsWz2tmShrpEgwFmY27nFaw0AOHDq3Mmn932QTQ4aQ@mail.gmail.com>
Message-ID: <CAA4nhyA25pmoBrN589-RRDfRLntOEnnTqDZkgP9NRHNYh=za=A@mail.gmail.com>

On Tue, Oct 4, 2011 at 8:56 AM, Juraj Fabo <juraj.fabo at gmail.com> wrote:

> Hi,
>
> I read through the previous answers several times again and did some
> progress.
> I found out that the valid codec_ms values are <10ms,60ms> and using lower
> values will not work.
>
> >> 80 means a hardware interrupt is received every 10ms with 80 bytes per
> >> time slot. This is the recommended mode. A value of 40 will increase
> >> interrupt load and not necesarily reduce your latency, you must reduce
> >> the "user period", which is how often the driver will deliver
> >> media/data to the user application. This is done using
> >> sangoma_tdm_set_usr_period().
> >>
> >>
> >> The codec_ms is used to call sangoma_tdm_set_usr_period(). This is how
> >> often the driver will deliver data to the user application (for
> >> example, waking it up from select()).
> >>
>
> In each of the the test below the codec_ms was set to 1 with hope
> of fastest data delivery, however this resulted to 20ms codec_ms
> which is default.
>
>
> > Please, have a look at the results with various parameters used:
> > mtu:80 txqueue_size=1  rxqueue_size=1   one-direction delay=60ms
> > round-trip delay=120ms
> > mtu:80 txqueue_size=2  rxqueue_size=2   one-direction delay=80ms
> > round-trip delay=160ms
> > mtu:80 txqueue_size=10 rxqueue_size=10 one-direction delay=240ms
> > round-trip delay=480ms
> > mtu:40 txqueue_size=1  rxqueue_size=1   one-direction delay=40ms
> > round-trip delay=80ms
> > mtu:40 txqueue_size=2  rxqueue_size=2   one-direction delay=60ms
> > round-trip delay=120ms
> > mtu:40 txqueue_size=5  rxqueue_size=5   one-direction delay=120ms
> > round-trip delay=240ms
> > mtu:16 txqueue_size=1  rxqueue_size=1   one-direction delay=30ms
> > round-trip delay=60ms
> > mtu:8  txqueue_size=1   rxqueue_size=1   one-direction delay=25ms
> > round-trip delay=50ms
> > mtu:8  txqueue_size=10 rxqueue_size=10 one-direction delay=205ms
> > round-trip delay=410ms
> >
>
> Updated test result with the codec_ms=10 is following:
> mtu:16 txqueue_size=1  rxqueue_size=10   one-direction delay=20ms
> round-trip delay=50ms
> mtu:80 txqueue_size=1  rxqueue_size=10   one-direction delay=56ms
> round-trip delay=110ms
>
> However, this is still not 10ms with the default MTU:80 and codec_ms:10 :(
>
> Anyway, is it a good idea to use so assymetric tx/rx queue sizes?
>
>
In general I don't think is a good idea because 99% of the user population
is using default symmetric queues, moving to obscure/rarely-tested
configurations increases your likelihood of facing problems. Having said
that, I don't see why it should not work :-)

Can you share the code you're using to measure latency? may be setup a git
repo in github with your code?

*Moises Silva
**Software Engineer, Development Manager***

msilva 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 x128

f.   +1 905 474 9223



**<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email+signatures>

Products<http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
 | Solutions<http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
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| YouTube<http://www.youtube.com/sangomatechnologies?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>

                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>
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From gvmello at yx.cl  Mon Oct 17 20:35:37 2011
From: gvmello at yx.cl (Glaucio Vaz de Mello)
Date: Mon, 17 Oct 2011 13:35:37 -0300
Subject: [Freeswitch-dev] FreeTDM delay reconnecting E1/T1
In-Reply-To: <CAA4nhyCOBei_aEmtDHqeFAYiSztNpYmncv993ZgwF744GGHYEg@mail.gmail.com>
References: <CAPLGU-71UcZxBD=M1CccVO0xYwNsRqH5Kk0KETyxZdmmHFtDvQ@mail.gmail.com>
	<CAA4nhyCs5rN54kUAz27RVCRY=fGGkshTHE96wtSt8oDFuF0bCg@mail.gmail.com>
	<CAPLGU-4xAjjgGWXtZHa+ukJ9vLZ0QtNz-knwmfoPc6gmngV6Aw@mail.gmail.com>
	<CAA4nhyCOBei_aEmtDHqeFAYiSztNpYmncv993ZgwF744GGHYEg@mail.gmail.com>
Message-ID: <CAPLGU-6SPvDB3=SyPddpDpjZ-1OaeAcX7YdhQPdJYxgyRKHNog@mail.gmail.com>

No, I'm not using the latest git head.
I'm going to upgrade and test it again. Thanks.



On Mon, Oct 17, 2011 at 1:16 PM, Moises Silva <moises.silva at gmail.com>wrote:

> On Mon, Oct 17, 2011 at 10:50 AM, Glaucio Vaz de Mello <gvmello at yx.cl>wrote:
>
>> Hi,
>>
>> Thanks for the help.
>> Yes, I tried ftdm dump and got the same results. This command was issued
>> after unplugging the cable:
>>
>> freeswitch at internal> ftdm dump wp1 1
>> span_id: 1
>> chan_id: 1
>> physical_span_id: 1
>> physical_chan_id: 1
>> physical_status: ok
>> signaling_status: UP
>> type: B
>> state: DOWN
>> last_state: DOWN
>> txgain: 0.00
>> rxgain: 0.00
>> cid_date:
>> cid_name:
>> cid_num:
>> ani:
>> aniII:
>> dnis:
>> rdnis:
>> cause: NONE
>> session: (none)
>>
>> -- No state history --
>>
>>
>>
> That 's odd. Are you using latest git HEAD? if you are using latest git
> HEAD, ping me on IRC and I can take a quick look. If is not solved within 10
> mins we'll have to open a JIRA ticket.
>
> *Moises Silva
> **Software Engineer, Development Manager***
>
> msilva 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 x128
>
> f.   +1 905 474 9223
>
>
>
> **<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email+signatures>
>
> Products<http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>  | Solutions<http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>  | Events<http://sangoma.com/about_us/events?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>  | Contact<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>  | Wiki<http://wiki.sangoma.com/?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
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>  | Twitter<http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>`|
> | YouTube<http://www.youtube.com/sangomatechnologies?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>
>                 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>
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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  Mon Oct 17 23:53:25 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 17 Oct 2011 12:53:25 -0700
Subject: [Freeswitch-dev] FreeSWITCH Conference Call Right Now! - Phil
	Zimmermann
In-Reply-To: <CAKq5mS7RwzNyN8O1UxXa4fXxOeEy7Tg-jMHHTtkwqchP+k-E7g@mail.gmail.com>
References: <CAKzWOxVekoCX+y961zkTkPD5VGFDHU-38C72OqYcW9QgcCZerA@mail.gmail.com>
	<CAKq5mS7RwzNyN8O1UxXa4fXxOeEy7Tg-jMHHTtkwqchP+k-E7g@mail.gmail.com>
Message-ID: <CAKzWOxU+xXMA8NsS6TtOsM6oyEef+8S2YGB++HOaeAWFd+mHYg@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Weekly_Conference_Call#Past_Calls

On Wed, Oct 12, 2011 at 5:13 PM, envelopes envelopes
<sunwood360 at gmail.com>wrote:

> Missed the event, is there a recorded archive?
> On Oct 12, 2011 10:06 AM, "Michael Collins" <msc at freeswitch.org> wrote:
>
>> Hey all,
>>
>> It seems my first email to the list didn't make it. Just a reminder that
>> Philip Zimmermann of PGP fame is coming in to talk about VoIP security, ZRTP
>> and Zfone. Please join us now!
>>
>> -Michael
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> 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
>>
>>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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 kheimerl at cs.berkeley.edu  Wed Oct 19 03:12:51 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Tue, 18 Oct 2011 16:12:51 -0700
Subject: [Freeswitch-dev] Calling Python Application from Chatplan (mod_sms
	help)
Message-ID: <CACyT-3mxBZQax2A94MHbOSXm6mwgAm_uD4fJAxqOi_iBRKrLag@mail.gmail.com>

I'm pushing this to FreeSwitch-dev as mod_sms is still under active
development.

Any way to call a Python Application from the chatplan?

This works:

<action application="set" data="test=${python(OpenBTS_Parse_SMS)}"/>

But I can't monkey with the session variables with the fs_api call.

Instead, I want to do this:

<action application="python" data="OpenBTS_Parse_SMS"/>

and this fails in the following way:

2011-10-18 16:12:02.776298 [ERR] switch_loadable_module.c:705 Invalid
chat application interface [python]!

The mod_sms page seems to indicate that lua works just fine from the
chatplan... is this just not implemented yet? Can I do that?

Thanks!


From kheimerl at cs.berkeley.edu  Wed Oct 19 04:00:18 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Tue, 18 Oct 2011 17:00:18 -0700
Subject: [Freeswitch-dev] Calling Python Application from Chatplan
	(mod_sms help)
In-Reply-To: <CACyT-3mxBZQax2A94MHbOSXm6mwgAm_uD4fJAxqOi_iBRKrLag@mail.gmail.com>
References: <CACyT-3mxBZQax2A94MHbOSXm6mwgAm_uD4fJAxqOi_iBRKrLag@mail.gmail.com>
Message-ID: <CACyT-3mWkP6mg8OS342K617SP6dsnhGX+9yM9vy2SsBYW_0qsw@mail.gmail.com>

Oh, and I can confirm that only Lua works from the chatplan.

On Tue, Oct 18, 2011 at 4:12 PM, Kurtis Heimerl
<kheimerl at cs.berkeley.edu> wrote:
> I'm pushing this to FreeSwitch-dev as mod_sms is still under active
> development.
>
> Any way to call a Python Application from the chatplan?
>
> This works:
>
> <action application="set" data="test=${python(OpenBTS_Parse_SMS)}"/>
>
> But I can't monkey with the session variables with the fs_api call.
>
> Instead, I want to do this:
>
> <action application="python" data="OpenBTS_Parse_SMS"/>
>
> and this fails in the following way:
>
> 2011-10-18 16:12:02.776298 [ERR] switch_loadable_module.c:705 Invalid
> chat application interface [python]!
>
> The mod_sms page seems to indicate that lua works just fine from the
> chatplan... is this just not implemented yet? Can I do that?
>
> Thanks!
>


From anthony.minessale at gmail.com  Wed Oct 19 19:23:44 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 19 Oct 2011 10:23:44 -0500
Subject: [Freeswitch-dev] Calling Python Application from Chatplan
 (mod_sms help)
In-Reply-To: <CACyT-3mWkP6mg8OS342K617SP6dsnhGX+9yM9vy2SsBYW_0qsw@mail.gmail.com>
References: <CACyT-3mxBZQax2A94MHbOSXm6mwgAm_uD4fJAxqOi_iBRKrLag@mail.gmail.com>
	<CACyT-3mWkP6mg8OS342K617SP6dsnhGX+9yM9vy2SsBYW_0qsw@mail.gmail.com>
Message-ID: <CAKbxfG89z+EWZn0+Zw44zJgbq52MQLxgACZFJ1_1-7zTDGnSHg@mail.gmail.com>

try latest

On Tue, Oct 18, 2011 at 7:00 PM, Kurtis Heimerl
<kheimerl at cs.berkeley.edu> wrote:
> Oh, and I can confirm that only Lua works from the chatplan.
>
> On Tue, Oct 18, 2011 at 4:12 PM, Kurtis Heimerl
> <kheimerl at cs.berkeley.edu> wrote:
>> I'm pushing this to FreeSwitch-dev as mod_sms is still under active
>> development.
>>
>> Any way to call a Python Application from the chatplan?
>>
>> This works:
>>
>> <action application="set" data="test=${python(OpenBTS_Parse_SMS)}"/>
>>
>> But I can't monkey with the session variables with the fs_api call.
>>
>> Instead, I want to do this:
>>
>> <action application="python" data="OpenBTS_Parse_SMS"/>
>>
>> and this fails in the following way:
>>
>> 2011-10-18 16:12:02.776298 [ERR] switch_loadable_module.c:705 Invalid
>> chat application interface [python]!
>>
>> The mod_sms page seems to indicate that lua works just fine from the
>> chatplan... is this just not implemented yet? Can I do that?
>>
>> Thanks!
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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 msc at freeswitch.org  Wed Oct 19 20:02:02 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 19 Oct 2011 09:02:02 -0700
Subject: [Freeswitch-dev] FreeSWITCH Conference Call Today
Message-ID: <CAKzWOxVddXvVRMUFDD3iL45T4zD=-sgfsfP+zRFK-N=9ArT8DQ@mail.gmail.com>

Hello all,

Our agenda today is light:
http://wiki.freeswitch.org/wiki/FS_weekly_2011_10_19

We're going to have a community scrum where we ask each other questions,
etc. I have a few minor updates for you all as well. Talk to you soon.

-Michael
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From kheimerl at cs.berkeley.edu  Wed Oct 19 20:57:05 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Wed, 19 Oct 2011 09:57:05 -0700
Subject: [Freeswitch-dev] Calling Python Application from Chatplan
 (mod_sms help)
In-Reply-To: <CAKbxfG89z+EWZn0+Zw44zJgbq52MQLxgACZFJ1_1-7zTDGnSHg@mail.gmail.com>
References: <CACyT-3mxBZQax2A94MHbOSXm6mwgAm_uD4fJAxqOi_iBRKrLag@mail.gmail.com>
	<CACyT-3mWkP6mg8OS342K617SP6dsnhGX+9yM9vy2SsBYW_0qsw@mail.gmail.com>
	<CAKbxfG89z+EWZn0+Zw44zJgbq52MQLxgACZFJ1_1-7zTDGnSHg@mail.gmail.com>
Message-ID: <CACyT-3nDD-SePsvj=6Sypxa+ZzC0S0rkrtL-HRqYTHHJd7JWBQ@mail.gmail.com>

Great, will do!

On Wed, Oct 19, 2011 at 8:23 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> try latest
>
> On Tue, Oct 18, 2011 at 7:00 PM, Kurtis Heimerl
> <kheimerl at cs.berkeley.edu> wrote:
>> Oh, and I can confirm that only Lua works from the chatplan.
>>
>> On Tue, Oct 18, 2011 at 4:12 PM, Kurtis Heimerl
>> <kheimerl at cs.berkeley.edu> wrote:
>>> I'm pushing this to FreeSwitch-dev as mod_sms is still under active
>>> development.
>>>
>>> Any way to call a Python Application from the chatplan?
>>>
>>> This works:
>>>
>>> <action application="set" data="test=${python(OpenBTS_Parse_SMS)}"/>
>>>
>>> But I can't monkey with the session variables with the fs_api call.
>>>
>>> Instead, I want to do this:
>>>
>>> <action application="python" data="OpenBTS_Parse_SMS"/>
>>>
>>> and this fails in the following way:
>>>
>>> 2011-10-18 16:12:02.776298 [ERR] switch_loadable_module.c:705 Invalid
>>> chat application interface [python]!
>>>
>>> The mod_sms page seems to indicate that lua works just fine from the
>>> chatplan... is this just not implemented yet? Can I do that?
>>>
>>> Thanks!
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> 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
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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 admharris at gmx.com  Wed Oct 19 23:21:15 2011
From: admharris at gmx.com (adam harris)
Date: Wed, 19 Oct 2011 21:21:15 +0200
Subject: [Freeswitch-dev] Couple of questions about esl in C
Message-ID: <20111019192116.297650@gmx.com>

A few questions about esl in C

1, After sending api playback command I want to wait until the file is done playing. How do I do this is C

2, Whats the proper way to use esl_recv_event_timed and esl_recv_even I tried it like this esl_recv_event_timed(call_info->global_handle, 10, 0, NULL); but that printed the complete event on screen. I want to do this in a while loop to check for event-name





Can you please resend this to freeswitch-dev at lists.freeswitch.org
-MC

On Wed, Oct 19, 2011 at 9:10 AM, adam harris <admharris at gmx.com> wrote:

 ? ?After sending ?api playback command I want to wait until the file is done playing

 ? ?How do I do this is C



From admharris at gmx.com  Wed Oct 19 23:37:06 2011
From: admharris at gmx.com (adam harris)
Date: Wed, 19 Oct 2011 21:37:06 +0200
Subject: [Freeswitch-dev] get variable value
Message-ID: <20111019193706.297660@gmx.com>

How do I get the vaule of a custom variable. I have seen uuid_getvar Usage: uuid_getvar <uuid> <varname> but how I use in C


From msc at freeswitch.org  Thu Oct 20 02:27:52 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 19 Oct 2011 15:27:52 -0700
Subject: [Freeswitch-dev] get variable value
In-Reply-To: <20111019193706.297660@gmx.com>
References: <20111019193706.297660@gmx.com>
Message-ID: <CAKzWOxV8z=sfVp__d4bL1ThVkr_Hq6R8UZ8h+WCEMRgS7GmtKg@mail.gmail.com>

Grep the source for "switch_channel_get_variable" and you'll see numerous
examples...

-MC

On Wed, Oct 19, 2011 at 12:37 PM, adam harris <admharris at gmx.com> wrote:

> How do I get the vaule of a custom variable. I have seen uuid_getvar Usage:
> uuid_getvar <uuid> <varname> but how I use in C
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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  Thu Oct 20 02:30:42 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 19 Oct 2011 15:30:42 -0700
Subject: [Freeswitch-dev] Couple of questions about esl in C
In-Reply-To: <20111019192116.297650@gmx.com>
References: <20111019192116.297650@gmx.com>
Message-ID: <CAKzWOxW+4h7oVJVg1bGNfYsOrNNGuvL5EmsBX6DUNZRfK+Ozfw@mail.gmail.com>

On Wed, Oct 19, 2011 at 12:21 PM, adam harris <admharris at gmx.com> wrote:

> A few questions about esl in C
>
> 1, After sending api playback command I want to wait until the file is done
> playing. How do I do this is C


> 2, Whats the proper way to use esl_recv_event_timed and esl_recv_even I
> tried it like this esl_recv_event_timed(call_info->global_handle, 10, 0,
> NULL); but that printed the complete event on screen. I want to do this in a
> while loop to check for event-name
>
> Would you mind putting some code snippets here or in
pastebin.freeswitch.org?
-MC
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From jmesquita at freeswitch.org  Thu Oct 20 02:55:19 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 19 Oct 2011 19:55:19 -0300
Subject: [Freeswitch-dev] get variable value
In-Reply-To: <CAKzWOxV8z=sfVp__d4bL1ThVkr_Hq6R8UZ8h+WCEMRgS7GmtKg@mail.gmail.com>
References: <20111019193706.297660@gmx.com>
	<CAKzWOxV8z=sfVp__d4bL1ThVkr_Hq6R8UZ8h+WCEMRgS7GmtKg@mail.gmail.com>
Message-ID: <CABH8QX3f4Gt6Z-us6yVaPJ5wOyVBetu2TMt93yt82W_ypp2ruw@mail.gmail.com>

>From your email, I am assuming you are trying to get a variable using ESL?
If so, custom variables are prefixed with variable. So you would use the
get_header functions just as you would on other non-custom variables.

Makes sense?

Regards,
Jo?o Mesquita



On Wed, Oct 19, 2011 at 7:27 PM, Michael Collins <msc at freeswitch.org> wrote:

> Grep the source for "switch_channel_get_variable" and you'll see numerous
> examples...
>
> -MC
>
> On Wed, Oct 19, 2011 at 12:37 PM, adam harris <admharris at gmx.com> wrote:
>
>> How do I get the vaule of a custom variable. I have seen uuid_getvar
>> Usage: uuid_getvar <uuid> <varname> but how I use in C
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> 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
>>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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  Thu Oct 20 10:34:38 2011
From: gabe at gundy.org (Gabriel Gunderson)
Date: Thu, 20 Oct 2011 00:34:38 -0600
Subject: [Freeswitch-dev] FreeSWITCH dialstring/channel variables
In-Reply-To: <CA+B+5Z+pL7=LtE9gx78XnLNMbeGOCip=7MSJmfyzrR3ySKkz9w@mail.gmail.com>
References: <CA+B+5Z+pL7=LtE9gx78XnLNMbeGOCip=7MSJmfyzrR3ySKkz9w@mail.gmail.com>
Message-ID: <CAMwZ-tL-xMyjBi6tC1fC3oW-r4BmVKvYNGJ-7gapLwKRtXXYXw@mail.gmail.com>

On Mon, Oct 10, 2011 at 8:37 AM, lg <lg at perfectitchina.com> wrote:
> One quick question here. Is it any dialstring to control dial function? Like
> as,i would change the dial?legs sequence as dial caller number first
> and?then when caller pickup the call,?dial the called destination number (
> as the same as "originate dial_string destination_number" function). Your
> hint or short response would be much appreciated.

I'm not sure I understand the question, but I *think* you're looking
for a bridge.


DIALPLAN:
This is how you would do it from the dialplan:

<action application="bridge"
data="sofia/internal/9998881111 at sip.yourprovider.com"/>

More on the Wiki:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge#From_the_Dialplan


COMMANDS:
Or, if you want to use commands, try something like this:

originate sofia/example/300 at foo.com &bridge(sofia/example/400 at bar.com)

More on the Wiki:
http://wiki.freeswitch.org/wiki/Mod_commands#originate


Good luck!

Gabe


From admharris at gmx.com  Thu Oct 20 04:48:22 2011
From: admharris at gmx.com (adam harris)
Date: Thu, 20 Oct 2011 02:48:22 +0200
Subject: [Freeswitch-dev] get variable value
Message-ID: <20111020004822.297630@gmx.com>


Yes I am using ESL but don't ?get_header functions need to catch an event first.
I was thinking if there was already a builtin function.

I can't remember exacty because it was a while ago but I used something like this in ruby

@con.api("uuid_getvar", "#{@uuid} myvar").getBody

> ----- Original Message -----
> From: Jo?o Mesquita
> Sent: 10/19/11 10:55 PM
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] get variable value
> 
> >From your email, I am assuming you are trying to get a variable using ESL?
> If so, custom variables are prefixed with variable. So you would use the
> get_header functions just as you would on other non-custom variables.
> 
> Makes sense?
> 
> Regards,
> Jo?o Mesquita
> 
> 
> 
> On Wed, Oct 19, 2011 at 7:27 PM, Michael Collins <msc at freeswitch.org> wrote:
> 
> > Grep the source for "switch_channel_get_variable" and you'll see numerous
> > examples...
> >
> > -MC
> >
> > On Wed, Oct 19, 2011 at 12:37 PM, adam harris <admharris at gmx.com> wrote:
> >
> >> How do I get the vaule of a custom variable. I have seen uuid_getvar
> >> Usage: uuid_getvar <uuid> <varname> but how I use in C
> >>
> >> _______________________________________________
> >> Join us at ClueCon 2011, Aug 9-11, Chicago
> >> http://www.cluecon.com 877-7-4ACLUE
> >>
> >> 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
> >>
> >
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > 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 admharris at gmx.com  Thu Oct 20 05:02:43 2011
From: admharris at gmx.com (adam harris)
Date: Thu, 20 Oct 2011 03:02:43 +0200
Subject: [Freeswitch-dev] Couple of questions about esl in C
Message-ID: <20111020010243.297650@gmx.com>


I want to try one more thing to see if it solves my problem

int response_timeout = (10 * 1000);
esl_status_t status;
status = esl_recv_event_timed(call_info->global_handle, response_timeout, 0, xxxxx); 

The note for the last paramter is:
\param[out] save_event If this is not NULL, will return the event received

What type of varible do I need to pass as the paramter and where will the return data go and how to access it.


I have noticed this channel variable
Application-Response: FILE%20PLAYED
which possibly might indicate playback completion?


> ----- Original Message -----
> From: Michael Collins
> Sent: 10/19/11 10:30 PM
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] Couple of questions about esl in C
> 
> On Wed, Oct 19, 2011 at 12:21 PM, adam harris <admharris at gmx.com> wrote:
> 
> > A few questions about esl in C
> >
> > 1, After sending api playback command I want to wait until the file is done
> > playing. How do I do this is C
> 
> 
> > 2, Whats the proper way to use esl_recv_event_timed and esl_recv_even I
> > tried it like this esl_recv_event_timed(call_info->global_handle, 10, 0,
> > NULL); but that printed the complete event on screen. I want to do this in a
> > while loop to check for event-name
> >
> > Would you mind putting some code snippets here or in
> pastebin.freeswitch.org?
> -MC



From jmesquita at freeswitch.org  Thu Oct 20 16:38:35 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Thu, 20 Oct 2011 09:38:35 -0300
Subject: [Freeswitch-dev] get variable value
In-Reply-To: <20111020004822.297630@gmx.com>
References: <20111020004822.297630@gmx.com>
Message-ID: <CABH8QX0Y9QEti0n3NOcizZU2vajFDBGF4wLUFSifDLYB-heYQA@mail.gmail.com>

The snippet you've sent is a whole different thing than C. You kinda have to
know these things to code on either language...

Using Ruby, the following statement DOES return an event:

@con.api("uuid_getvar", "#{@uuid} myvar")

And with that event, you implicitly call "getBody".

When you are using C, you can't use these things like that. You have to
explicitly make function calls to the correct event. Look at the ESL
examples testserver.c and testclient.c on the source and maybe you'll
understand.

When calling api on C, you need to use the handle->last_sr_reply (if I
remember correctly).

I hope that helps, otherwise, a working code would.

Regards,
Jo?o Mesquita



On Wed, Oct 19, 2011 at 9:48 PM, adam harris <admharris at gmx.com> wrote:

>
> Yes I am using ESL but don't  get_header functions need to catch an event
> first.
> I was thinking if there was already a builtin function.
>
> I can't remember exacty because it was a while ago but I used something
> like this in ruby
>
> @con.api("uuid_getvar", "#{@uuid} myvar").getBody
>
> > ----- Original Message -----
> > From: Jo?o Mesquita
> > Sent: 10/19/11 10:55 PM
> > To: freeswitch-dev at lists.freeswitch.org
> > Subject: Re: [Freeswitch-dev] get variable value
> >
> > >From your email, I am assuming you are trying to get a variable using
> ESL?
> > If so, custom variables are prefixed with variable. So you would use the
> > get_header functions just as you would on other non-custom variables.
> >
> > Makes sense?
> >
> > Regards,
> > Jo?o Mesquita
> >
> >
> >
> > On Wed, Oct 19, 2011 at 7:27 PM, Michael Collins <msc at freeswitch.org>
> wrote:
> >
> > > Grep the source for "switch_channel_get_variable" and you'll see
> numerous
> > > examples...
> > >
> > > -MC
> > >
> > > On Wed, Oct 19, 2011 at 12:37 PM, adam harris <admharris at gmx.com>
> wrote:
> > >
> > >> How do I get the vaule of a custom variable. I have seen uuid_getvar
> > >> Usage: uuid_getvar <uuid> <varname> but how I use in C
> > >>
> > >> _______________________________________________
> > >> Join us at ClueCon 2011, Aug 9-11, Chicago
> > >> http://www.cluecon.com 877-7-4ACLUE
> > >>
> > >> 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
> > >>
> > >
> > >
> > > _______________________________________________
> > > Join us at ClueCon 2011, Aug 9-11, Chicago
> > > http://www.cluecon.com 877-7-4ACLUE
> > >
> > > 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
> > >
> > >
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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 gvmello at yx.cl  Thu Oct 20 18:21:09 2011
From: gvmello at yx.cl (Glaucio Vaz de Mello)
Date: Thu, 20 Oct 2011 11:21:09 -0300
Subject: [Freeswitch-dev] FreeTDM delay reconnecting E1/T1
In-Reply-To: <CAPLGU-6SPvDB3=SyPddpDpjZ-1OaeAcX7YdhQPdJYxgyRKHNog@mail.gmail.com>
References: <CAPLGU-71UcZxBD=M1CccVO0xYwNsRqH5Kk0KETyxZdmmHFtDvQ@mail.gmail.com>
	<CAA4nhyCs5rN54kUAz27RVCRY=fGGkshTHE96wtSt8oDFuF0bCg@mail.gmail.com>
	<CAPLGU-4xAjjgGWXtZHa+ukJ9vLZ0QtNz-knwmfoPc6gmngV6Aw@mail.gmail.com>
	<CAA4nhyCOBei_aEmtDHqeFAYiSztNpYmncv993ZgwF744GGHYEg@mail.gmail.com>
	<CAPLGU-6SPvDB3=SyPddpDpjZ-1OaeAcX7YdhQPdJYxgyRKHNog@mail.gmail.com>
Message-ID: <CAPLGU-4B2oyU+i6B0d09=GULe0kUViYaYW1EVPSotR45oAr0ig@mail.gmail.com>

Hello,

Just to tell you that I tested it again with latest Git source and latest
sangoma driver.
Delay still happens, but are less than before.

When E1/T1 cable is disconnected, wanpipemon command shows "Loss of Signal"
in less than 1 second.
Meanwhile, fs_cli "ftdm dump" still shows "physical status" as "ok" for 4 or
5 seconds, then it turns to "alarmed".

Reconnecting the cable I experienced bigger delays: while wanpipemon shows
"Loss of Signal OFF" in less than 1 second, fs_cli "ftdm dump" takes more
than 20 seconds to became "ok" again.

If you think this should be fixed and need more information, logs or ssh
access, just let me know and I'll be glad to provide it.


regards,
Glaucio




On Mon, Oct 17, 2011 at 1:35 PM, Glaucio Vaz de Mello <gvmello at yx.cl> wrote:

> No, I'm not using the latest git head.
> I'm going to upgrade and test it again. Thanks.
>
>
>
> On Mon, Oct 17, 2011 at 1:16 PM, Moises Silva <moises.silva at gmail.com>wrote:
>
>> On Mon, Oct 17, 2011 at 10:50 AM, Glaucio Vaz de Mello <gvmello at yx.cl>wrote:
>>
>>> Hi,
>>>
>>> Thanks for the help.
>>> Yes, I tried ftdm dump and got the same results. This command was issued
>>> after unplugging the cable:
>>>
>>> freeswitch at internal> ftdm dump wp1 1
>>> span_id: 1
>>> chan_id: 1
>>> physical_span_id: 1
>>> physical_chan_id: 1
>>> physical_status: ok
>>> signaling_status: UP
>>> type: B
>>> state: DOWN
>>> last_state: DOWN
>>> txgain: 0.00
>>> rxgain: 0.00
>>> cid_date:
>>> cid_name:
>>> cid_num:
>>> ani:
>>> aniII:
>>> dnis:
>>> rdnis:
>>> cause: NONE
>>> session: (none)
>>>
>>> -- No state history --
>>>
>>>
>>>
>> That 's odd. Are you using latest git HEAD? if you are using latest git
>> HEAD, ping me on IRC and I can take a quick look. If is not solved within 10
>> mins we'll have to open a JIRA ticket.
>>
>> *Moises Silva
>> **Software Engineer, Development Manager***
>>
>> msilva 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 x128
>>
>> f.   +1 905 474 9223
>>
>>
>>
>> **<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email+signatures>
>>
>> Products<http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>>  | Solutions<http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>>  | Events<http://sangoma.com/about_us/events?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>>  | Contact<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>>  | Wiki<http://wiki.sangoma.com/?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>>  | Facebook<http://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>>  | Twitter<http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>`|
>> | YouTube<http://www.youtube.com/sangomatechnologies?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>>
>>                 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>
>>
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> 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 xin at ind.rwth-aachen.de  Thu Oct 20 20:09:24 2011
From: xin at ind.rwth-aachen.de (Han Xin)
Date: Thu, 20 Oct 2011 18:09:24 +0200
Subject: [Freeswitch-dev] trouble on how to remove the media bug from
 the session after the sessions are closed
In-Reply-To: <CAKbxfG9sRo3fVNAR+_+BoSv6rYsEhv7OKba8QVs8Yvsv7oR4_Q@mail.gmail.com>
Message-ID: <314ebfaf9a5bf1bbd700ce3ba8a7920b@gw.ind.rwth-aachen.de>

 I tried a lot in the past days but still got no progress. So I upload my code here to ask for help again.

Thanks anyway!
 
----------------original message-----------------
From: "Anthony Minessale" anthony.minessale at gmail.com 
To: freeswitch-dev at lists.freeswitch.org 
Date: Mon, 17 Oct 2011 09:39:26 -0500
-------------------------------------------------
 
 
> I have no more guesses. You need to review your custom code and find
> out where you read locked your session.
> 
> 
> On Mon, Oct 17, 2011 at 2:38 AM, Han Xin xin at ind.rwth-aachen.de wrote:
>> no, no use of ivr_originate function.
>> I constructed a helper struct which include the remote session, could this 
>> cause the session releasing problem?
>>
>> ----------------urspr?ngliche Nachricht-----------------
>> Von: "Anthony Minessale" anthony.minessale at gmail.com 
>> An: freeswitch-dev at lists.freeswitch.org 
>> Datum: Thu, 13 Oct 2011 15:55:35 -0500
>> -------------------------------------------------
>>
>>
>>> how about switch_ivr_originate() ?
>>>
>>> On Thu, Oct 13, 2011 at 2:19 AM, Han Xin xin at ind.rwth-aachen.de wrote:
>>>> yes I use a custom mod and the latest git on Oct 10 but I did not use the 
>>>> function
>>>> "switch_core_session_locate".
>>>>
>>>> ----------------urspr?ngliche Nachricht-----------------
>>>> Von: "Anthony Minessale" anthony.minessale at gmail.com 
>>>> An: freeswitch-dev at lists.freeswitch.org 
>>>> Datum: Wed, 12 Oct 2011 17:06:41 -0500
>>>> -------------------------------------------------
>>>>
>>>>
>>>>> looks like you have some custom mods maybe you used
>>>>> switch_core_session_locate and did not rwunlock it?
>>>>> are you using latest GIT ?
>>>>>
>>>>>
>>>>> On Mon, Oct 10, 2011 at 11:03 AM, Han Xin xin at ind.rwth-aachen.de 
>>>>> wrote:
>>>>>> yes, I use the SWITCH_ABC_TYPE_CLOSE type and I've found the 
>>>>>> media
>>>>>> bugs are
>>>>>> automatically removed with the DEBUG log infomation. Now the 
>>>>>> problem
>>>>>> is the
>>>>>> session ending.
>>>>>>
>>>>>> Detailed situation description: I added two media bugs to both 
>>>>>> local
>>>>>> session
>>>>>> and remote session of one call. When I ended the call, the media 
>>>>>> bugs are
>>>>>> removed
>>>>>> first, and then the session should be ended. But only the local 
>>>>>> session
>>>>>> is ended
>>>>>> correctly.
>>>>>> When I press F3 and F4 to show current channels and calls, the 
>>>>>> remote
>>>>>> channel
>>>>>> and call info is till there. But when I try to dump uuid to get 
>>>>>> channel
>>>>>> infomation
>>>>>> with the cmd "uuid_dump remote_session_uuid", it says there 
>>>>>> is no
>>>>>> such
>>>>>> channel!
>>>>>>
>>>>>> here is some related log info:
>>>>>>
>>>>>> 2011-10-10 17:36:41.428463 [ERR] mod_bwe.c:345
>>>>>> 28b3a542-aecd-4b9b-9f55-3c2674a25b5c: Remove local 
>>>>>> variable
>>>>>> store!
>>>>>> 2011-10-10 17:36:41.428463 [DEBUG] 
>>>>>> switch_ivr_bridge.c:1332
>>>>>> sofia/internal/sip:1001 at 137.226.198.252 :5060 skip 
>>>>>> receive
>>>>>> message
>>>>>> [UNBRIDGE] (channel is hungup already)
>>>>>> 2011-10-10 17:36:41.428463 [DEBUG] 
>>>>>> switch_ivr_bridge.c:1335
>>>>>> sofia/internal/1000 at asterisk skip receive message 
>>>>>> [UNBRIDGE]
>>>>>> (channel is
>>>>>> hungup already)
>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>> switch_core_media_bug.c:480
>>>>>> Removing BUG from 
>>>>>> sofia/internal/sip:1001 at 137.226.198.252 
>>>>>> :5060
>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>> switch_core_state_machine.c:602
>>>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State 
>>>>>> HANGUP
>>>>>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:454
>>>>>> sofia/internal/sip:1001 at 137.226.198.252 :5060 
>>>>>> Overriding SIP
>>>>>> cause 480
>>>>>> with 200 from the other leg
>>>>>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:460 
>>>>>> Channel
>>>>>> sofia/internal/sip:1001 at 137.226.198.252 :5060 hanging 
>>>>>> up,
>>>>>> cause:
>>>>>> NORMAL_CLEARING
>>>>>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:503 
>>>>>> Sending BYE
>>>>>> to
>>>>>> sofia/internal/sip:1001 at 137.226.198.252 :5060
>>>>>> 2011-10-10 17:36:41.428463 [DEBUG] 
>>>>>> switch_core_session.c:2235
>>>>>> sofia/internal/1000 at asterisk skip receive message
>>>>>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>> switch_core_state_machine.c:417
>>>>>> (sofia/internal/1000 at asterisk ) State EXECUTE going to 
>>>>>> sleep
>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>> switch_core_state_machine.c:362
>>>>>> (sofia/internal/1000 at asterisk ) Running State Change 
>>>>>> CS_HANGUP
>>>>>> 2011-10-10 17:36:41.428463 [ERR] mod_bwe.c:345
>>>>>> 713d5f0c-4341-47ec-b0bf-4b14bd6b4726: Remove local 
>>>>>> variable
>>>>>> store!
>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>> switch_core_media_bug.c:480
>>>>>> Removing BUG from sofia/internal/1000 at asterisk 
>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>> switch_core_state_machine.c:602
>>>>>> (sofia/internal/1000 at asterisk ) State HANGUP
>>>>>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:454
>>>>>> sofia/internal/1000 at asterisk Overriding SIP cause 480 with 
>>>>>> 200
>>>>>> from the
>>>>>> other leg
>>>>>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:460 
>>>>>> Channel
>>>>>> sofia/internal/1000 at asterisk hanging up, cause:
>>>>>> NORMAL_CLEARING
>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>> switch_core_state_machine.c:47
>>>>>> sofia/internal/1000 at asterisk Standard HANGUP, cause:
>>>>>> NORMAL_CLEARING
>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>> switch_core_state_machine.c:47
>>>>>> sofia/internal/sip:1001 at 137.226.198.252 :5060 Standard 
>>>>>> HANGUP,
>>>>>> cause:
>>>>>> NORMAL_CLEARING
>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>> switch_core_state_machine.c:602
>>>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State 
>>>>>> HANGUP
>>>>>> going to
>>>>>> sleep
>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>> switch_core_state_machine.c:393
>>>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State 
>>>>>> Change
>>>>>> CS_HANGUP
>>>>>> -> CS_REPORTING
>>>>>> 2011-10-10 17:36:41.428463 [DEBUG] 
>>>>>> switch_core_session.c:1177
>>>>>> Send
>>>>>> signal sofia/internal/sip:1001 at 137.226.198.252 :5060 
>>>>>> [BREAK]
>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>> switch_core_state_machine.c:362
>>>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) Running 
>>>>>> State
>>>>>> Change
>>>>>> CS_REPORTING
>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>> switch_core_state_machine.c:662
>>>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State
>>>>>> REPORTING
>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>> switch_core_state_machine.c:79
>>>>>> sofia/internal/sip:1001 at 137.226.198.252 :5060 Standard
>>>>>> REPORTING,
>>>>>> cause: NORMAL_CLEARING
>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>> switch_core_state_machine.c:662
>>>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State
>>>>>> REPORTING going
>>>>>> to sleep
>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>> switch_core_state_machine.c:387
>>>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State 
>>>>>> Change
>>>>>> CS_REPORTING -> CS_DESTROY
>>>>>> 2011-10-10 17:36:41.428463 [DEBUG] 
>>>>>> switch_core_session.c:1177
>>>>>> Send
>>>>>> signal sofia/internal/sip:1001 at 137.226.198.252 :5060 
>>>>>> [BREAK]
>>>>>> 2011-10-10 17:36:41.428463 [DEBUG] 
>>>>>> switch_core_session.c:1351
>>>>>> Session
>>>>>> 8 (sofia/internal/sip:1001 at 137.226.198.252 :5060) 
>>>>>> Locked,
>>>>>> Waiting on
>>>>>> external entities
>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>> switch_core_state_machine.c:602
>>>>>> (sofia/internal/1000 at asterisk ) State HANGUP going to 
>>>>>> sleep
>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>> switch_core_state_machine.c:393
>>>>>> (sofia/internal/1000 at asterisk ) State Change CS_HANGUP ->
>>>>>> CS_REPORTING
>>>>>> 2011-10-10 17:36:41.428463 [DEBUG] 
>>>>>> switch_core_session.c:1177
>>>>>> Send
>>>>>> signal sofia/internal/1000 at asterisk [BREAK]
>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>> switch_core_state_machine.c:362
>>>>>> (sofia/internal/1000 at asterisk ) Running State Change
>>>>>> CS_REPORTING
>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>> switch_core_state_machine.c:662
>>>>>> (sofia/internal/1000 at asterisk ) State REPORTING
>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>> switch_core_state_machine.c:79
>>>>>> sofia/internal/1000 at asterisk Standard REPORTING, cause:
>>>>>> NORMAL_CLEARING
>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>> switch_core_state_machine.c:662
>>>>>> (sofia/internal/1000 at asterisk ) State REPORTING going to 
>>>>>> sleep
>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>> switch_core_state_machine.c:387
>>>>>> (sofia/internal/1000 at asterisk ) State Change CS_REPORTING 
>>>>>> ->
>>>>>> CS_DESTROY
>>>>>> 2011-10-10 17:36:41.428463 [DEBUG] 
>>>>>> switch_core_session.c:1177
>>>>>> Send
>>>>>> signal sofia/internal/1000 at asterisk [BREAK]
>>>>>> 2011-10-10 17:36:41.428463 [DEBUG] 
>>>>>> switch_core_session.c:1351
>>>>>> Session
>>>>>> 7 (sofia/internal/1000 at asterisk ) Locked, Waiting on 
>>>>>> external
>>>>>> entities
>>>>>> 2011-10-10 17:36:41.428463 [NOTICE]
>>>>>> switch_core_session.c:1369
>>>>>> Session 7 (sofia/internal/1000 at asterisk ) Ended
>>>>>> 2011-10-10 17:36:41.428463 [NOTICE] 
>>>>>> switch_core_session.c:1371
>>>>>> Close
>>>>>> Channel sofia/internal/1000 at asterisk [CS_DESTROY]
>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>> switch_core_state_machine.c:491
>>>>>> (sofia/internal/1000 at asterisk ) Callstate Change HANGUP -> 
>>>>>> DOWN
>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>> switch_core_state_machine.c:494
>>>>>> (sofia/internal/1000 at asterisk ) Running State Change 
>>>>>> CS_DESTROY
>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>> switch_core_state_machine.c:504
>>>>>> (sofia/internal/1000 at asterisk ) State DESTROY
>>>>>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:365
>>>>>> sofia/internal/1000 at asterisk SOFIA DESTROY
>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>> switch_core_state_machine.c:86
>>>>>> sofia/internal/1000 at asterisk Standard DESTROY
>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>> switch_core_state_machine.c:504
>>>>>> (sofia/internal/1000 at asterisk ) State DESTROY going to 
>>>>>> sleep
>>>>>>
>>>>>> Could anyone give me some hints why the remote session is not 
>>>>>> ended
>>>>>> automatically?
>>>>>>
>>>>>>
>>>>>> ----------------urspr?ngliche 
>>>>>> Nachricht-----------------
>>>>>> Von: "Anthony Minessale" anthony.minessale at gmail.com 
>>>>>> An: freeswitch-dev at lists.freeswitch.org 
>>>>>> Datum: Mon, 10 Oct 2011 10:17:36 -0500
>>>>>> -------------------------------------------------
>>>>>>
>>>>>>
>>>>>>> it calls the callback with type set to 
>>>>>>> SWITCH_ABC_TYPE_CLOSE
>>>>>>>
>>>>>>> On Wed, Oct 5, 2011 at 7:37 PM, Vu Quang Hoa 
>>>>>>> vutamhoan at gmail.com 
>>>>>>> wrote:
>>>>>>>> As I tested - FS automatically removes bug when call 
>>>>>>>> ends.
>>>>>>>> I'm trying to figure out if it does the same with 
>>>>>>>> allocated
>>>>>>>> resources
>>>>>>>> in
>>>>>>>> add_bug routine.
>>>>>>>>
>>>>>>>> On Tue, Oct 4, 2011 at 7:03 PM, Han Xin 
>>>>>>>> xin at ind.rwth-aachen.de 
>>>>>>>> wrote:
>>>>>>>>>
>>>>>>>>> Hello everyone!
>>>>>>>>>
>>>>>>>>> I am new to Freeswitch and try to add media bugs such 
>>>>>>>>> as
>>>>>>>>> digital
>>>>>>>>> signal
>>>>>>>>> processing to the read stream and write stream. But I 
>>>>>>>>> have
>>>>>>>>> some
>>>>>>>>> trouble on
>>>>>>>>> how to remove the media bug from the session after 
>>>>>>>>> the
>>>>>>>>> sessions are
>>>>>>>>> closed.
>>>>>>>>> I tried to look into other existed modules such as
>>>>>>>>> mod_soundtouch
>>>>>>>>> and
>>>>>>>>> mod_dptools to find a solution, the former 
>>>>>>>>> mod_soundtouch
>>>>>>>>> put
>>>>>>>>> the
>>>>>>>>> function
>>>>>>>>> "switch_core_media_bug_remove()" in the 
>>>>>>>>> beginning of
>>>>>>>>> application
>>>>>>>>> run
>>>>>>>>> function. The latter mod use a single app called 
>>>>>>>>> "remove
>>>>>>>>> bugs" and
>>>>>>>>> put
>>>>>>>>> this
>>>>>>>>> app in the dial plan to remove the bugs.
>>>>>>>>>
>>>>>>>>> The question is where should I put this
>>>>>>>>> "switch_core_media_bug_remove()"
>>>>>>>>> function after the sessions are closed.
>>>>>>>>> Is there a way to automatically remove a media bug 
>>>>>>>>> when a call
>>>>>>>>> is
>>>>>>>>> ended?
>>>>>>>>> Or it is only possible to use a single app to remove the 
>>>>>>>>> media
>>>>>>>>> bugs as
>>>>>>>>> in
>>>>>>>>> the mod_dptools?
>>>>>>>>>
>>>>>>>>> Thank you all in advance!
>>>>>>>>>
>>>>>>>>> --
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> 
>>>>>>>>> _______________________________________________
>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>
>>>>>>>>> FreeSWITCH-dev mailing list
>>>>>>>>> FreeSWITCH-dev at lists.freeswitch.org 
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> 
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswi
>>>>>>>>> tch-
>>>>>>>>> dev
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> 
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opti
>>>>>>>>> ons/
>>>>>>>>> free
>>>>>>>>> swit
>>>>>>>>> ch-dev
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>> 
>>>>>>>> _______________________________________________
>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>
>>>>>>>> FreeSWITCH-dev mailing list
>>>>>>>> FreeSWITCH-dev at lists.freeswitch.org 
>>>>>>>>
>>>>>>>>
>>>>>>>> 
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitc
>>>>>>>> h-de
>>>>>>>> v
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> 
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/option
>>>>>>>> s/fr
>>>>>>>> eesw
>>>>>>>> itch
>>>>>>>> -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
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>
>>>>>>> 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/
>>>>>>> free
>>>>>>> swit
>>>>>>> ch-d
>>>>>>> ev
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>> --
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> FreeSWITCH-dev mailing list
>>>>>> FreeSWITCH-dev at lists.freeswitch.org 
>>>>>>
>>>>>> 
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-de
>>>>>> v
>>>>>>
>>>>>>
>>>>>> 
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/fr
>>>>>> eesw
>>>>>> itch
>>>>>> -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
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> 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/free
>>>>> swit
>>>>> ch-d
>>>>> ev
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>> --
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> 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/freesw
>>>> itch
>>>> -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
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> 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/freeswit
>>> ch-d
>>> ev
>>> http://www.freeswitch.org
>>>
>>
>> --
>>
>>
>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> 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
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> 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-d
> ev
> http://www.freeswitch.org
> 

-- 

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From anthony.minessale at gmail.com  Thu Oct 20 21:45:29 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 20 Oct 2011 12:45:29 -0500
Subject: [Freeswitch-dev] trouble on how to remove the media bug from
 the session after the sessions are closed
In-Reply-To: <314ebfaf9a5bf1bbd700ce3ba8a7920b@gw.ind.rwth-aachen.de>
References: <CAKbxfG9sRo3fVNAR+_+BoSv6rYsEhv7OKba8QVs8Yvsv7oR4_Q@mail.gmail.com>
	<314ebfaf9a5bf1bbd700ce3ba8a7920b@gw.ind.rwth-aachen.de>
Message-ID: <CAKbxfG-ch_ea--_Ve20Uih_7AzKWPQw80hCpeBadWGGkNqsFyA@mail.gmail.com>

try getting a gdb trace while the app is running, maybe the interface
you connected to is blocking?

On Thu, Oct 20, 2011 at 11:09 AM, Han Xin <xin at ind.rwth-aachen.de> wrote:
> ?I tried a lot in the past days but still got no progress. So I upload my code here to ask for help again.
>
> Thanks anyway!
>
> ----------------original message-----------------
> From: "Anthony Minessale" anthony.minessale at gmail.com
> To: freeswitch-dev at lists.freeswitch.org
> Date: Mon, 17 Oct 2011 09:39:26 -0500
> -------------------------------------------------
>
>
>> I have no more guesses. You need to review your custom code and find
>> out where you read locked your session.
>>
>>
>> On Mon, Oct 17, 2011 at 2:38 AM, Han Xin xin at ind.rwth-aachen.de wrote:
>>> no, no use of ivr_originate function.
>>> I constructed a helper struct which include the remote session, could this
>>> cause the session releasing problem?
>>>
>>> ----------------urspr?ngliche Nachricht-----------------
>>> Von: "Anthony Minessale" anthony.minessale at gmail.com
>>> An: freeswitch-dev at lists.freeswitch.org
>>> Datum: Thu, 13 Oct 2011 15:55:35 -0500
>>> -------------------------------------------------
>>>
>>>
>>>> how about switch_ivr_originate() ?
>>>>
>>>> On Thu, Oct 13, 2011 at 2:19 AM, Han Xin xin at ind.rwth-aachen.de wrote:
>>>>> yes I use a custom mod and the latest git on Oct 10 but I did not use the
>>>>> function
>>>>> "switch_core_session_locate".
>>>>>
>>>>> ----------------urspr?ngliche Nachricht-----------------
>>>>> Von: "Anthony Minessale" anthony.minessale at gmail.com
>>>>> An: freeswitch-dev at lists.freeswitch.org
>>>>> Datum: Wed, 12 Oct 2011 17:06:41 -0500
>>>>> -------------------------------------------------
>>>>>
>>>>>
>>>>>> looks like you have some custom mods maybe you used
>>>>>> switch_core_session_locate and did not rwunlock it?
>>>>>> are you using latest GIT ?
>>>>>>
>>>>>>
>>>>>> On Mon, Oct 10, 2011 at 11:03 AM, Han Xin xin at ind.rwth-aachen.de
>>>>>> wrote:
>>>>>>> yes, I use the SWITCH_ABC_TYPE_CLOSE type and I've found the
>>>>>>> media
>>>>>>> bugs are
>>>>>>> automatically removed with the DEBUG log infomation. Now the
>>>>>>> problem
>>>>>>> is the
>>>>>>> session ending.
>>>>>>>
>>>>>>> Detailed situation description: I added two media bugs to both
>>>>>>> local
>>>>>>> session
>>>>>>> and remote session of one call. When I ended the call, the media
>>>>>>> bugs are
>>>>>>> removed
>>>>>>> first, and then the session should be ended. But only the local
>>>>>>> session
>>>>>>> is ended
>>>>>>> correctly.
>>>>>>> When I press F3 and F4 to show current channels and calls, the
>>>>>>> remote
>>>>>>> channel
>>>>>>> and call info is till there. But when I try to dump uuid to get
>>>>>>> channel
>>>>>>> infomation
>>>>>>> with the cmd "uuid_dump remote_session_uuid", it says there
>>>>>>> is no
>>>>>>> such
>>>>>>> channel!
>>>>>>>
>>>>>>> here is some related log info:
>>>>>>>
>>>>>>> 2011-10-10 17:36:41.428463 [ERR] mod_bwe.c:345
>>>>>>> 28b3a542-aecd-4b9b-9f55-3c2674a25b5c: Remove local
>>>>>>> variable
>>>>>>> store!
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_ivr_bridge.c:1332
>>>>>>> sofia/internal/sip:1001 at 137.226.198.252 :5060 skip
>>>>>>> receive
>>>>>>> message
>>>>>>> [UNBRIDGE] (channel is hungup already)
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_ivr_bridge.c:1335
>>>>>>> sofia/internal/1000 at asterisk skip receive message
>>>>>>> [UNBRIDGE]
>>>>>>> (channel is
>>>>>>> hungup already)
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_media_bug.c:480
>>>>>>> Removing BUG from
>>>>>>> sofia/internal/sip:1001 at 137.226.198.252
>>>>>>> :5060
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:602
>>>>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State
>>>>>>> HANGUP
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:454
>>>>>>> sofia/internal/sip:1001 at 137.226.198.252 :5060
>>>>>>> Overriding SIP
>>>>>>> cause 480
>>>>>>> with 200 from the other leg
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:460
>>>>>>> Channel
>>>>>>> sofia/internal/sip:1001 at 137.226.198.252 :5060 hanging
>>>>>>> up,
>>>>>>> cause:
>>>>>>> NORMAL_CLEARING
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:503
>>>>>>> Sending BYE
>>>>>>> to
>>>>>>> sofia/internal/sip:1001 at 137.226.198.252 :5060
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_session.c:2235
>>>>>>> sofia/internal/1000 at asterisk skip receive message
>>>>>>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:417
>>>>>>> (sofia/internal/1000 at asterisk ) State EXECUTE going to
>>>>>>> sleep
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:362
>>>>>>> (sofia/internal/1000 at asterisk ) Running State Change
>>>>>>> CS_HANGUP
>>>>>>> 2011-10-10 17:36:41.428463 [ERR] mod_bwe.c:345
>>>>>>> 713d5f0c-4341-47ec-b0bf-4b14bd6b4726: Remove local
>>>>>>> variable
>>>>>>> store!
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_media_bug.c:480
>>>>>>> Removing BUG from sofia/internal/1000 at asterisk
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:602
>>>>>>> (sofia/internal/1000 at asterisk ) State HANGUP
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:454
>>>>>>> sofia/internal/1000 at asterisk Overriding SIP cause 480 with
>>>>>>> 200
>>>>>>> from the
>>>>>>> other leg
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:460
>>>>>>> Channel
>>>>>>> sofia/internal/1000 at asterisk hanging up, cause:
>>>>>>> NORMAL_CLEARING
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:47
>>>>>>> sofia/internal/1000 at asterisk Standard HANGUP, cause:
>>>>>>> NORMAL_CLEARING
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:47
>>>>>>> sofia/internal/sip:1001 at 137.226.198.252 :5060 Standard
>>>>>>> HANGUP,
>>>>>>> cause:
>>>>>>> NORMAL_CLEARING
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:602
>>>>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State
>>>>>>> HANGUP
>>>>>>> going to
>>>>>>> sleep
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:393
>>>>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State
>>>>>>> Change
>>>>>>> CS_HANGUP
>>>>>>> -> CS_REPORTING
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_session.c:1177
>>>>>>> Send
>>>>>>> signal sofia/internal/sip:1001 at 137.226.198.252 :5060
>>>>>>> [BREAK]
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:362
>>>>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) Running
>>>>>>> State
>>>>>>> Change
>>>>>>> CS_REPORTING
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:662
>>>>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State
>>>>>>> REPORTING
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:79
>>>>>>> sofia/internal/sip:1001 at 137.226.198.252 :5060 Standard
>>>>>>> REPORTING,
>>>>>>> cause: NORMAL_CLEARING
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:662
>>>>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State
>>>>>>> REPORTING going
>>>>>>> to sleep
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:387
>>>>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State
>>>>>>> Change
>>>>>>> CS_REPORTING -> CS_DESTROY
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_session.c:1177
>>>>>>> Send
>>>>>>> signal sofia/internal/sip:1001 at 137.226.198.252 :5060
>>>>>>> [BREAK]
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_session.c:1351
>>>>>>> Session
>>>>>>> 8 (sofia/internal/sip:1001 at 137.226.198.252 :5060)
>>>>>>> Locked,
>>>>>>> Waiting on
>>>>>>> external entities
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:602
>>>>>>> (sofia/internal/1000 at asterisk ) State HANGUP going to
>>>>>>> sleep
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:393
>>>>>>> (sofia/internal/1000 at asterisk ) State Change CS_HANGUP ->
>>>>>>> CS_REPORTING
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_session.c:1177
>>>>>>> Send
>>>>>>> signal sofia/internal/1000 at asterisk [BREAK]
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:362
>>>>>>> (sofia/internal/1000 at asterisk ) Running State Change
>>>>>>> CS_REPORTING
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:662
>>>>>>> (sofia/internal/1000 at asterisk ) State REPORTING
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:79
>>>>>>> sofia/internal/1000 at asterisk Standard REPORTING, cause:
>>>>>>> NORMAL_CLEARING
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:662
>>>>>>> (sofia/internal/1000 at asterisk ) State REPORTING going to
>>>>>>> sleep
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:387
>>>>>>> (sofia/internal/1000 at asterisk ) State Change CS_REPORTING
>>>>>>> ->
>>>>>>> CS_DESTROY
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_session.c:1177
>>>>>>> Send
>>>>>>> signal sofia/internal/1000 at asterisk [BREAK]
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_session.c:1351
>>>>>>> Session
>>>>>>> 7 (sofia/internal/1000 at asterisk ) Locked, Waiting on
>>>>>>> external
>>>>>>> entities
>>>>>>> 2011-10-10 17:36:41.428463 [NOTICE]
>>>>>>> switch_core_session.c:1369
>>>>>>> Session 7 (sofia/internal/1000 at asterisk ) Ended
>>>>>>> 2011-10-10 17:36:41.428463 [NOTICE]
>>>>>>> switch_core_session.c:1371
>>>>>>> Close
>>>>>>> Channel sofia/internal/1000 at asterisk [CS_DESTROY]
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:491
>>>>>>> (sofia/internal/1000 at asterisk ) Callstate Change HANGUP ->
>>>>>>> DOWN
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:494
>>>>>>> (sofia/internal/1000 at asterisk ) Running State Change
>>>>>>> CS_DESTROY
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:504
>>>>>>> (sofia/internal/1000 at asterisk ) State DESTROY
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:365
>>>>>>> sofia/internal/1000 at asterisk SOFIA DESTROY
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:86
>>>>>>> sofia/internal/1000 at asterisk Standard DESTROY
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:504
>>>>>>> (sofia/internal/1000 at asterisk ) State DESTROY going to
>>>>>>> sleep
>>>>>>>
>>>>>>> Could anyone give me some hints why the remote session is not
>>>>>>> ended
>>>>>>> automatically?
>>>>>>>
>>>>>>>
>>>>>>> ----------------urspr?ngliche
>>>>>>> Nachricht-----------------
>>>>>>> Von: "Anthony Minessale" anthony.minessale at gmail.com
>>>>>>> An: freeswitch-dev at lists.freeswitch.org
>>>>>>> Datum: Mon, 10 Oct 2011 10:17:36 -0500
>>>>>>> -------------------------------------------------
>>>>>>>
>>>>>>>
>>>>>>>> it calls the callback with type set to
>>>>>>>> SWITCH_ABC_TYPE_CLOSE
>>>>>>>>
>>>>>>>> On Wed, Oct 5, 2011 at 7:37 PM, Vu Quang Hoa
>>>>>>>> vutamhoan at gmail.com
>>>>>>>> wrote:
>>>>>>>>> As I tested - FS automatically removes bug when call
>>>>>>>>> ends.
>>>>>>>>> I'm trying to figure out if it does the same with
>>>>>>>>> allocated
>>>>>>>>> resources
>>>>>>>>> in
>>>>>>>>> add_bug routine.
>>>>>>>>>
>>>>>>>>> On Tue, Oct 4, 2011 at 7:03 PM, Han Xin
>>>>>>>>> xin at ind.rwth-aachen.de
>>>>>>>>> wrote:
>>>>>>>>>>
>>>>>>>>>> Hello everyone!
>>>>>>>>>>
>>>>>>>>>> I am new to Freeswitch and try to add media bugs such
>>>>>>>>>> as
>>>>>>>>>> digital
>>>>>>>>>> signal
>>>>>>>>>> processing to the read stream and write stream. But I
>>>>>>>>>> have
>>>>>>>>>> some
>>>>>>>>>> trouble on
>>>>>>>>>> how to remove the media bug from the session after
>>>>>>>>>> the
>>>>>>>>>> sessions are
>>>>>>>>>> closed.
>>>>>>>>>> I tried to look into other existed modules such as
>>>>>>>>>> mod_soundtouch
>>>>>>>>>> and
>>>>>>>>>> mod_dptools to find a solution, the former
>>>>>>>>>> mod_soundtouch
>>>>>>>>>> put
>>>>>>>>>> the
>>>>>>>>>> function
>>>>>>>>>> "switch_core_media_bug_remove()" in the
>>>>>>>>>> beginning of
>>>>>>>>>> application
>>>>>>>>>> run
>>>>>>>>>> function. The latter mod use a single app called
>>>>>>>>>> "remove
>>>>>>>>>> bugs" and
>>>>>>>>>> put
>>>>>>>>>> this
>>>>>>>>>> app in the dial plan to remove the bugs.
>>>>>>>>>>
>>>>>>>>>> The question is where should I put this
>>>>>>>>>> "switch_core_media_bug_remove()"
>>>>>>>>>> function after the sessions are closed.
>>>>>>>>>> Is there a way to automatically remove a media bug
>>>>>>>>>> when a call
>>>>>>>>>> is
>>>>>>>>>> ended?
>>>>>>>>>> Or it is only possible to use a single app to remove the
>>>>>>>>>> media
>>>>>>>>>> bugs as
>>>>>>>>>> in
>>>>>>>>>> the mod_dptools?
>>>>>>>>>>
>>>>>>>>>> Thank you all in advance!
>>>>>>>>>>
>>>>>>>>>> --
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>
>>>>>>>>>> FreeSWITCH-dev mailing list
>>>>>>>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswi
>>>>>>>>>> tch-
>>>>>>>>>> dev
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opti
>>>>>>>>>> ons/
>>>>>>>>>> free
>>>>>>>>>> swit
>>>>>>>>>> ch-dev
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>
>>>>>>>>> FreeSWITCH-dev mailing list
>>>>>>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitc
>>>>>>>>> h-de
>>>>>>>>> v
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/option
>>>>>>>>> s/fr
>>>>>>>>> eesw
>>>>>>>>> itch
>>>>>>>>> -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
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>
>>>>>>>> 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/
>>>>>>>> free
>>>>>>>> swit
>>>>>>>> ch-d
>>>>>>>> ev
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>
>>>>>>> FreeSWITCH-dev mailing list
>>>>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-de
>>>>>>> v
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/fr
>>>>>>> eesw
>>>>>>> itch
>>>>>>> -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
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> 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/free
>>>>>> swit
>>>>>> ch-d
>>>>>> ev
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>> --
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> 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/freesw
>>>>> itch
>>>>> -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
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> 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/freeswit
>>>> ch-d
>>>> ev
>>>> http://www.freeswitch.org
>>>>
>>>
>>> --
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> 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
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> 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-d
>> ev
>> http://www.freeswitch.org
>>
>
> --
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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 kheimerl at cs.berkeley.edu  Fri Oct 21 00:40:39 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Thu, 20 Oct 2011 13:40:39 -0700
Subject: [Freeswitch-dev] Calling Python Application from Chatplan
 (mod_sms help)
In-Reply-To: <CACyT-3nDD-SePsvj=6Sypxa+ZzC0S0rkrtL-HRqYTHHJd7JWBQ@mail.gmail.com>
References: <CACyT-3mxBZQax2A94MHbOSXm6mwgAm_uD4fJAxqOi_iBRKrLag@mail.gmail.com>
	<CACyT-3mWkP6mg8OS342K617SP6dsnhGX+9yM9vy2SsBYW_0qsw@mail.gmail.com>
	<CAKbxfG89z+EWZn0+Zw44zJgbq52MQLxgACZFJ1_1-7zTDGnSHg@mail.gmail.com>
	<CACyT-3nDD-SePsvj=6Sypxa+ZzC0S0rkrtL-HRqYTHHJd7JWBQ@mail.gmail.com>
Message-ID: <CACyT-3mYZAA66dx7yLX5rdDhs-1Hcm+GbuW6HiypaUPXBcHSwA@mail.gmail.com>

Still not working:

    <extension name="openbts">
      <condition field="to_user" expression="^smsc$">
	<action application="set" data="api_result=${log(${to})}"/>
	<action application="python" data="OpenBTS_Parse_SMS"/>
      </condition>
    </extension>

2011-10-20 13:39:49.534714 [INFO] mod_sms.c:300 Processing text
message IMSI641104278340879->smsc in context default
2011-10-20 13:39:49.534714 [ERR] switch_loadable_module.c:705 Invalid
chat application interface [python]!

On Wed, Oct 19, 2011 at 9:57 AM, Kurtis Heimerl
<kheimerl at cs.berkeley.edu> wrote:
> Great, will do!
>
> On Wed, Oct 19, 2011 at 8:23 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> try latest
>>
>> On Tue, Oct 18, 2011 at 7:00 PM, Kurtis Heimerl
>> <kheimerl at cs.berkeley.edu> wrote:
>>> Oh, and I can confirm that only Lua works from the chatplan.
>>>
>>> On Tue, Oct 18, 2011 at 4:12 PM, Kurtis Heimerl
>>> <kheimerl at cs.berkeley.edu> wrote:
>>>> I'm pushing this to FreeSwitch-dev as mod_sms is still under active
>>>> development.
>>>>
>>>> Any way to call a Python Application from the chatplan?
>>>>
>>>> This works:
>>>>
>>>> <action application="set" data="test=${python(OpenBTS_Parse_SMS)}"/>
>>>>
>>>> But I can't monkey with the session variables with the fs_api call.
>>>>
>>>> Instead, I want to do this:
>>>>
>>>> <action application="python" data="OpenBTS_Parse_SMS"/>
>>>>
>>>> and this fails in the following way:
>>>>
>>>> 2011-10-18 16:12:02.776298 [ERR] switch_loadable_module.c:705 Invalid
>>>> chat application interface [python]!
>>>>
>>>> The mod_sms page seems to indicate that lua works just fine from the
>>>> chatplan... is this just not implemented yet? Can I do that?
>>>>
>>>> Thanks!
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> 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
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> 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 Oct 21 01:37:23 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 20 Oct 2011 16:37:23 -0500
Subject: [Freeswitch-dev] Calling Python Application from Chatplan
 (mod_sms help)
In-Reply-To: <CACyT-3mYZAA66dx7yLX5rdDhs-1Hcm+GbuW6HiypaUPXBcHSwA@mail.gmail.com>
References: <CACyT-3mxBZQax2A94MHbOSXm6mwgAm_uD4fJAxqOi_iBRKrLag@mail.gmail.com>
	<CACyT-3mWkP6mg8OS342K617SP6dsnhGX+9yM9vy2SsBYW_0qsw@mail.gmail.com>
	<CAKbxfG89z+EWZn0+Zw44zJgbq52MQLxgACZFJ1_1-7zTDGnSHg@mail.gmail.com>
	<CACyT-3nDD-SePsvj=6Sypxa+ZzC0S0rkrtL-HRqYTHHJd7JWBQ@mail.gmail.com>
	<CACyT-3mYZAA66dx7yLX5rdDhs-1Hcm+GbuW6HiypaUPXBcHSwA@mail.gmail.com>
Message-ID: <CAKbxfG8PibyP-ymp63miAD0GjO5ytJZ4xi88c7BeWHwRm=UpzQ@mail.gmail.com>

from the code:

SWITCH_ADD_CHAT_APP(chat_app_interface, "python", "execute a python
script", "execute a python script", python_chat_function, "<script>",
SCAF_NONE);

you must not have the version loaded that implemented it.


On Thu, Oct 20, 2011 at 3:40 PM, Kurtis Heimerl
<kheimerl at cs.berkeley.edu> wrote:
> Still not working:
>
> ? ?<extension name="openbts">
> ? ? ?<condition field="to_user" expression="^smsc$">
> ? ? ? ?<action application="set" data="api_result=${log(${to})}"/>
> ? ? ? ?<action application="python" data="OpenBTS_Parse_SMS"/>
> ? ? ?</condition>
> ? ?</extension>
>
> 2011-10-20 13:39:49.534714 [INFO] mod_sms.c:300 Processing text
> message IMSI641104278340879->smsc in context default
> 2011-10-20 13:39:49.534714 [ERR] switch_loadable_module.c:705 Invalid
> chat application interface [python]!
>
> On Wed, Oct 19, 2011 at 9:57 AM, Kurtis Heimerl
> <kheimerl at cs.berkeley.edu> wrote:
>> Great, will do!
>>
>> On Wed, Oct 19, 2011 at 8:23 AM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>>> try latest
>>>
>>> On Tue, Oct 18, 2011 at 7:00 PM, Kurtis Heimerl
>>> <kheimerl at cs.berkeley.edu> wrote:
>>>> Oh, and I can confirm that only Lua works from the chatplan.
>>>>
>>>> On Tue, Oct 18, 2011 at 4:12 PM, Kurtis Heimerl
>>>> <kheimerl at cs.berkeley.edu> wrote:
>>>>> I'm pushing this to FreeSwitch-dev as mod_sms is still under active
>>>>> development.
>>>>>
>>>>> Any way to call a Python Application from the chatplan?
>>>>>
>>>>> This works:
>>>>>
>>>>> <action application="set" data="test=${python(OpenBTS_Parse_SMS)}"/>
>>>>>
>>>>> But I can't monkey with the session variables with the fs_api call.
>>>>>
>>>>> Instead, I want to do this:
>>>>>
>>>>> <action application="python" data="OpenBTS_Parse_SMS"/>
>>>>>
>>>>> and this fails in the following way:
>>>>>
>>>>> 2011-10-18 16:12:02.776298 [ERR] switch_loadable_module.c:705 Invalid
>>>>> chat application interface [python]!
>>>>>
>>>>> The mod_sms page seems to indicate that lua works just fine from the
>>>>> chatplan... is this just not implemented yet? Can I do that?
>>>>>
>>>>> Thanks!
>>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> 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
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> 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
>>>
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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 kheimerl at cs.berkeley.edu  Fri Oct 21 06:25:54 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Thu, 20 Oct 2011 19:25:54 -0700
Subject: [Freeswitch-dev] Calling Python Application from Chatplan
 (mod_sms help)
In-Reply-To: <CAKbxfG8PibyP-ymp63miAD0GjO5ytJZ4xi88c7BeWHwRm=UpzQ@mail.gmail.com>
References: <CACyT-3mxBZQax2A94MHbOSXm6mwgAm_uD4fJAxqOi_iBRKrLag@mail.gmail.com>
	<CACyT-3mWkP6mg8OS342K617SP6dsnhGX+9yM9vy2SsBYW_0qsw@mail.gmail.com>
	<CAKbxfG89z+EWZn0+Zw44zJgbq52MQLxgACZFJ1_1-7zTDGnSHg@mail.gmail.com>
	<CACyT-3nDD-SePsvj=6Sypxa+ZzC0S0rkrtL-HRqYTHHJd7JWBQ@mail.gmail.com>
	<CACyT-3mYZAA66dx7yLX5rdDhs-1Hcm+GbuW6HiypaUPXBcHSwA@mail.gmail.com>
	<CAKbxfG8PibyP-ymp63miAD0GjO5ytJZ4xi88c7BeWHwRm=UpzQ@mail.gmail.com>
Message-ID: <CACyT-3=2gk=7hXAca-QAm27s-1f0eefJrz=v3xrfbhmZUS2ksw@mail.gmail.com>

Sorry about that, you're right. For some reason, python wasn't
building with my main install. Got that remedied and everything seems
to be working! Thanks!

One quick question, is there any way to access session variables
(e.g., ${domain}, or anything from vars.xml) from the "message" object
that freeswitch hands a chat application?

On Thu, Oct 20, 2011 at 2:37 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> from the code:
>
> SWITCH_ADD_CHAT_APP(chat_app_interface, "python", "execute a python
> script", "execute a python script", python_chat_function, "<script>",
> SCAF_NONE);
>
> you must not have the version loaded that implemented it.
>
>
> On Thu, Oct 20, 2011 at 3:40 PM, Kurtis Heimerl
> <kheimerl at cs.berkeley.edu> wrote:
>> Still not working:
>>
>> ? ?<extension name="openbts">
>> ? ? ?<condition field="to_user" expression="^smsc$">
>> ? ? ? ?<action application="set" data="api_result=${log(${to})}"/>
>> ? ? ? ?<action application="python" data="OpenBTS_Parse_SMS"/>
>> ? ? ?</condition>
>> ? ?</extension>
>>
>> 2011-10-20 13:39:49.534714 [INFO] mod_sms.c:300 Processing text
>> message IMSI641104278340879->smsc in context default
>> 2011-10-20 13:39:49.534714 [ERR] switch_loadable_module.c:705 Invalid
>> chat application interface [python]!
>>
>> On Wed, Oct 19, 2011 at 9:57 AM, Kurtis Heimerl
>> <kheimerl at cs.berkeley.edu> wrote:
>>> Great, will do!
>>>
>>> On Wed, Oct 19, 2011 at 8:23 AM, Anthony Minessale
>>> <anthony.minessale at gmail.com> wrote:
>>>> try latest
>>>>
>>>> On Tue, Oct 18, 2011 at 7:00 PM, Kurtis Heimerl
>>>> <kheimerl at cs.berkeley.edu> wrote:
>>>>> Oh, and I can confirm that only Lua works from the chatplan.
>>>>>
>>>>> On Tue, Oct 18, 2011 at 4:12 PM, Kurtis Heimerl
>>>>> <kheimerl at cs.berkeley.edu> wrote:
>>>>>> I'm pushing this to FreeSwitch-dev as mod_sms is still under active
>>>>>> development.
>>>>>>
>>>>>> Any way to call a Python Application from the chatplan?
>>>>>>
>>>>>> This works:
>>>>>>
>>>>>> <action application="set" data="test=${python(OpenBTS_Parse_SMS)}"/>
>>>>>>
>>>>>> But I can't monkey with the session variables with the fs_api call.
>>>>>>
>>>>>> Instead, I want to do this:
>>>>>>
>>>>>> <action application="python" data="OpenBTS_Parse_SMS"/>
>>>>>>
>>>>>> and this fails in the following way:
>>>>>>
>>>>>> 2011-10-18 16:12:02.776298 [ERR] switch_loadable_module.c:705 Invalid
>>>>>> chat application interface [python]!
>>>>>>
>>>>>> The mod_sms page seems to indicate that lua works just fine from the
>>>>>> chatplan... is this just not implemented yet? Can I do that?
>>>>>>
>>>>>> Thanks!
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> 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
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> 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
>>>>
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> 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
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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 mrene_lists at avgs.ca  Fri Oct 21 07:39:33 2011
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Thu, 20 Oct 2011 23:39:33 -0400
Subject: [Freeswitch-dev] trouble on how to remove the media bug from
	the session after the sessions are closed
In-Reply-To: <314ebfaf9a5bf1bbd700ce3ba8a7920b@gw.ind.rwth-aachen.de>
References: <314ebfaf9a5bf1bbd700ce3ba8a7920b@gw.ind.rwth-aachen.de>
Message-ID: <E8A95E8A-E508-4ED1-8CFB-D195DC27C8AB@avgs.ca>

Hi,

On line 188, you call switch_core_session_get_partner. That function internally calls switch_core_session_locate(), you have to rwunlock it.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 2011-10-20, at 12:09 PM, Han Xin wrote:

> I tried a lot in the past days but still got no progress. So I upload my code here to ask for help again.
> 
> Thanks anyway!
> 
> ----------------original message-----------------
> From: "Anthony Minessale" anthony.minessale at gmail.com
> To: freeswitch-dev at lists.freeswitch.org
> Date: Mon, 17 Oct 2011 09:39:26 -0500
> -------------------------------------------------
> 
> 
>> I have no more guesses. You need to review your custom code and find
>> out where you read locked your session.
>> 
>> 
>> On Mon, Oct 17, 2011 at 2:38 AM, Han Xin xin at ind.rwth-aachen.de wrote:
>>> no, no use of ivr_originate function.
>>> I constructed a helper struct which include the remote session, could this
>>> cause the session releasing problem?
>>> 
>>> ----------------urspr?ngliche Nachricht-----------------
>>> Von: "Anthony Minessale" anthony.minessale at gmail.com
>>> An: freeswitch-dev at lists.freeswitch.org
>>> Datum: Thu, 13 Oct 2011 15:55:35 -0500
>>> -------------------------------------------------
>>> 
>>> 
>>>> how about switch_ivr_originate() ?
>>>> 
>>>> On Thu, Oct 13, 2011 at 2:19 AM, Han Xin xin at ind.rwth-aachen.de wrote:
>>>>> yes I use a custom mod and the latest git on Oct 10 but I did not use the
>>>>> function
>>>>> "switch_core_session_locate".
>>>>> 
>>>>> ----------------urspr?ngliche Nachricht-----------------
>>>>> Von: "Anthony Minessale" anthony.minessale at gmail.com
>>>>> An: freeswitch-dev at lists.freeswitch.org
>>>>> Datum: Wed, 12 Oct 2011 17:06:41 -0500
>>>>> -------------------------------------------------
>>>>> 
>>>>> 
>>>>>> looks like you have some custom mods maybe you used
>>>>>> switch_core_session_locate and did not rwunlock it?
>>>>>> are you using latest GIT ?
>>>>>> 
>>>>>> 
>>>>>> On Mon, Oct 10, 2011 at 11:03 AM, Han Xin xin at ind.rwth-aachen.de
>>>>>> wrote:
>>>>>>> yes, I use the SWITCH_ABC_TYPE_CLOSE type and I've found the
>>>>>>> media
>>>>>>> bugs are
>>>>>>> automatically removed with the DEBUG log infomation. Now the
>>>>>>> problem
>>>>>>> is the
>>>>>>> session ending.
>>>>>>> 
>>>>>>> Detailed situation description: I added two media bugs to both
>>>>>>> local
>>>>>>> session
>>>>>>> and remote session of one call. When I ended the call, the media
>>>>>>> bugs are
>>>>>>> removed
>>>>>>> first, and then the session should be ended. But only the local
>>>>>>> session
>>>>>>> is ended
>>>>>>> correctly.
>>>>>>> When I press F3 and F4 to show current channels and calls, the
>>>>>>> remote
>>>>>>> channel
>>>>>>> and call info is till there. But when I try to dump uuid to get
>>>>>>> channel
>>>>>>> infomation
>>>>>>> with the cmd "uuid_dump remote_session_uuid", it says there
>>>>>>> is no
>>>>>>> such
>>>>>>> channel!
>>>>>>> 
>>>>>>> here is some related log info:
>>>>>>> 
>>>>>>> 2011-10-10 17:36:41.428463 [ERR] mod_bwe.c:345
>>>>>>> 28b3a542-aecd-4b9b-9f55-3c2674a25b5c: Remove local
>>>>>>> variable
>>>>>>> store!
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_ivr_bridge.c:1332
>>>>>>> sofia/internal/sip:1001 at 137.226.198.252 :5060 skip
>>>>>>> receive
>>>>>>> message
>>>>>>> [UNBRIDGE] (channel is hungup already)
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_ivr_bridge.c:1335
>>>>>>> sofia/internal/1000 at asterisk skip receive message
>>>>>>> [UNBRIDGE]
>>>>>>> (channel is
>>>>>>> hungup already)
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_media_bug.c:480
>>>>>>> Removing BUG from
>>>>>>> sofia/internal/sip:1001 at 137.226.198.252
>>>>>>> :5060
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:602
>>>>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State
>>>>>>> HANGUP
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:454
>>>>>>> sofia/internal/sip:1001 at 137.226.198.252 :5060
>>>>>>> Overriding SIP
>>>>>>> cause 480
>>>>>>> with 200 from the other leg
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:460
>>>>>>> Channel
>>>>>>> sofia/internal/sip:1001 at 137.226.198.252 :5060 hanging
>>>>>>> up,
>>>>>>> cause:
>>>>>>> NORMAL_CLEARING
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:503
>>>>>>> Sending BYE
>>>>>>> to
>>>>>>> sofia/internal/sip:1001 at 137.226.198.252 :5060
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_session.c:2235
>>>>>>> sofia/internal/1000 at asterisk skip receive message
>>>>>>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:417
>>>>>>> (sofia/internal/1000 at asterisk ) State EXECUTE going to
>>>>>>> sleep
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:362
>>>>>>> (sofia/internal/1000 at asterisk ) Running State Change
>>>>>>> CS_HANGUP
>>>>>>> 2011-10-10 17:36:41.428463 [ERR] mod_bwe.c:345
>>>>>>> 713d5f0c-4341-47ec-b0bf-4b14bd6b4726: Remove local
>>>>>>> variable
>>>>>>> store!
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_media_bug.c:480
>>>>>>> Removing BUG from sofia/internal/1000 at asterisk
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:602
>>>>>>> (sofia/internal/1000 at asterisk ) State HANGUP
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:454
>>>>>>> sofia/internal/1000 at asterisk Overriding SIP cause 480 with
>>>>>>> 200
>>>>>>> from the
>>>>>>> other leg
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:460
>>>>>>> Channel
>>>>>>> sofia/internal/1000 at asterisk hanging up, cause:
>>>>>>> NORMAL_CLEARING
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:47
>>>>>>> sofia/internal/1000 at asterisk Standard HANGUP, cause:
>>>>>>> NORMAL_CLEARING
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:47
>>>>>>> sofia/internal/sip:1001 at 137.226.198.252 :5060 Standard
>>>>>>> HANGUP,
>>>>>>> cause:
>>>>>>> NORMAL_CLEARING
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:602
>>>>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State
>>>>>>> HANGUP
>>>>>>> going to
>>>>>>> sleep
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:393
>>>>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State
>>>>>>> Change
>>>>>>> CS_HANGUP
>>>>>>> -> CS_REPORTING
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_session.c:1177
>>>>>>> Send
>>>>>>> signal sofia/internal/sip:1001 at 137.226.198.252 :5060
>>>>>>> [BREAK]
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:362
>>>>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) Running
>>>>>>> State
>>>>>>> Change
>>>>>>> CS_REPORTING
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:662
>>>>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State
>>>>>>> REPORTING
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:79
>>>>>>> sofia/internal/sip:1001 at 137.226.198.252 :5060 Standard
>>>>>>> REPORTING,
>>>>>>> cause: NORMAL_CLEARING
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:662
>>>>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State
>>>>>>> REPORTING going
>>>>>>> to sleep
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:387
>>>>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State
>>>>>>> Change
>>>>>>> CS_REPORTING -> CS_DESTROY
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_session.c:1177
>>>>>>> Send
>>>>>>> signal sofia/internal/sip:1001 at 137.226.198.252 :5060
>>>>>>> [BREAK]
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_session.c:1351
>>>>>>> Session
>>>>>>> 8 (sofia/internal/sip:1001 at 137.226.198.252 :5060)
>>>>>>> Locked,
>>>>>>> Waiting on
>>>>>>> external entities
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:602
>>>>>>> (sofia/internal/1000 at asterisk ) State HANGUP going to
>>>>>>> sleep
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:393
>>>>>>> (sofia/internal/1000 at asterisk ) State Change CS_HANGUP ->
>>>>>>> CS_REPORTING
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_session.c:1177
>>>>>>> Send
>>>>>>> signal sofia/internal/1000 at asterisk [BREAK]
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:362
>>>>>>> (sofia/internal/1000 at asterisk ) Running State Change
>>>>>>> CS_REPORTING
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:662
>>>>>>> (sofia/internal/1000 at asterisk ) State REPORTING
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:79
>>>>>>> sofia/internal/1000 at asterisk Standard REPORTING, cause:
>>>>>>> NORMAL_CLEARING
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:662
>>>>>>> (sofia/internal/1000 at asterisk ) State REPORTING going to
>>>>>>> sleep
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:387
>>>>>>> (sofia/internal/1000 at asterisk ) State Change CS_REPORTING
>>>>>>> ->
>>>>>>> CS_DESTROY
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_session.c:1177
>>>>>>> Send
>>>>>>> signal sofia/internal/1000 at asterisk [BREAK]
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_session.c:1351
>>>>>>> Session
>>>>>>> 7 (sofia/internal/1000 at asterisk ) Locked, Waiting on
>>>>>>> external
>>>>>>> entities
>>>>>>> 2011-10-10 17:36:41.428463 [NOTICE]
>>>>>>> switch_core_session.c:1369
>>>>>>> Session 7 (sofia/internal/1000 at asterisk ) Ended
>>>>>>> 2011-10-10 17:36:41.428463 [NOTICE]
>>>>>>> switch_core_session.c:1371
>>>>>>> Close
>>>>>>> Channel sofia/internal/1000 at asterisk [CS_DESTROY]
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:491
>>>>>>> (sofia/internal/1000 at asterisk ) Callstate Change HANGUP ->
>>>>>>> DOWN
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:494
>>>>>>> (sofia/internal/1000 at asterisk ) Running State Change
>>>>>>> CS_DESTROY
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:504
>>>>>>> (sofia/internal/1000 at asterisk ) State DESTROY
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:365
>>>>>>> sofia/internal/1000 at asterisk SOFIA DESTROY
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:86
>>>>>>> sofia/internal/1000 at asterisk Standard DESTROY
>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>> switch_core_state_machine.c:504
>>>>>>> (sofia/internal/1000 at asterisk ) State DESTROY going to
>>>>>>> sleep
>>>>>>> 
>>>>>>> Could anyone give me some hints why the remote session is not
>>>>>>> ended
>>>>>>> automatically?
>>>>>>> 
>>>>>>> 
>>>>>>> ----------------urspr?ngliche
>>>>>>> Nachricht-----------------
>>>>>>> Von: "Anthony Minessale" anthony.minessale at gmail.com
>>>>>>> An: freeswitch-dev at lists.freeswitch.org
>>>>>>> Datum: Mon, 10 Oct 2011 10:17:36 -0500
>>>>>>> -------------------------------------------------
>>>>>>> 
>>>>>>> 
>>>>>>>> it calls the callback with type set to
>>>>>>>> SWITCH_ABC_TYPE_CLOSE
>>>>>>>> 
>>>>>>>> On Wed, Oct 5, 2011 at 7:37 PM, Vu Quang Hoa
>>>>>>>> vutamhoan at gmail.com
>>>>>>>> wrote:
>>>>>>>>> As I tested - FS automatically removes bug when call
>>>>>>>>> ends.
>>>>>>>>> I'm trying to figure out if it does the same with
>>>>>>>>> allocated
>>>>>>>>> resources
>>>>>>>>> in
>>>>>>>>> add_bug routine.
>>>>>>>>> 
>>>>>>>>> On Tue, Oct 4, 2011 at 7:03 PM, Han Xin
>>>>>>>>> xin at ind.rwth-aachen.de
>>>>>>>>> wrote:
>>>>>>>>>> 
>>>>>>>>>> Hello everyone!
>>>>>>>>>> 
>>>>>>>>>> I am new to Freeswitch and try to add media bugs such
>>>>>>>>>> as
>>>>>>>>>> digital
>>>>>>>>>> signal
>>>>>>>>>> processing to the read stream and write stream. But I
>>>>>>>>>> have
>>>>>>>>>> some
>>>>>>>>>> trouble on
>>>>>>>>>> how to remove the media bug from the session after
>>>>>>>>>> the
>>>>>>>>>> sessions are
>>>>>>>>>> closed.
>>>>>>>>>> I tried to look into other existed modules such as
>>>>>>>>>> mod_soundtouch
>>>>>>>>>> and
>>>>>>>>>> mod_dptools to find a solution, the former
>>>>>>>>>> mod_soundtouch
>>>>>>>>>> put
>>>>>>>>>> the
>>>>>>>>>> function
>>>>>>>>>> "switch_core_media_bug_remove()" in the
>>>>>>>>>> beginning of
>>>>>>>>>> application
>>>>>>>>>> run
>>>>>>>>>> function. The latter mod use a single app called
>>>>>>>>>> "remove
>>>>>>>>>> bugs" and
>>>>>>>>>> put
>>>>>>>>>> this
>>>>>>>>>> app in the dial plan to remove the bugs.
>>>>>>>>>> 
>>>>>>>>>> The question is where should I put this
>>>>>>>>>> "switch_core_media_bug_remove()"
>>>>>>>>>> function after the sessions are closed.
>>>>>>>>>> Is there a way to automatically remove a media bug
>>>>>>>>>> when a call
>>>>>>>>>> is
>>>>>>>>>> ended?
>>>>>>>>>> Or it is only possible to use a single app to remove the
>>>>>>>>>> media
>>>>>>>>>> bugs as
>>>>>>>>>> in
>>>>>>>>>> the mod_dptools?
>>>>>>>>>> 
>>>>>>>>>> Thank you all in advance!
>>>>>>>>>> 
>>>>>>>>>> --
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> _______________________________________________
>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>> 
>>>>>>>>>> FreeSWITCH-dev mailing list
>>>>>>>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswi
>>>>>>>>>> tch-
>>>>>>>>>> dev
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opti
>>>>>>>>>> ons/
>>>>>>>>>> free
>>>>>>>>>> swit
>>>>>>>>>> ch-dev
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>> _______________________________________________
>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>> 
>>>>>>>>> FreeSWITCH-dev mailing list
>>>>>>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitc
>>>>>>>>> h-de
>>>>>>>>> v
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/option
>>>>>>>>> s/fr
>>>>>>>>> eesw
>>>>>>>>> itch
>>>>>>>>> -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
>>>>>>>> 
>>>>>>>> _______________________________________________
>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>> 
>>>>>>>> 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/
>>>>>>>> free
>>>>>>>> swit
>>>>>>>> ch-d
>>>>>>>> ev
>>>>>>>> http://www.freeswitch.org
>>>>>>>> 
>>>>>>> 
>>>>>>> --
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>> 
>>>>>>> FreeSWITCH-dev mailing list
>>>>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>>>>> 
>>>>>>> 
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-de
>>>>>>> v
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/fr
>>>>>>> eesw
>>>>>>> itch
>>>>>>> -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
>>>>>> 
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>> 
>>>>>> 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/free
>>>>>> swit
>>>>>> ch-d
>>>>>> ev
>>>>>> http://www.freeswitch.org
>>>>>> 
>>>>> 
>>>>> --
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>> 
>>>>> 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/freesw
>>>>> itch
>>>>> -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
>>>> 
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>> 
>>>> FreeSWITCH-dev mailing list
>>>> FreeSWITCH-dev at lists.freeswitch.org
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>>>> 
>>>> 
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>>>> ch-d
>>>> ev
>>>> http://www.freeswitch.org
>>>> 
>>> 
>>> --
>>> 
>>> 
>>> 
>>> 
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
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>> 
>> 
>> 
>> --
>> Anthony Minessale II
>> 
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>> 
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
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>> IRC: irc.freenode.net #freeswitch
>> 
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>> 
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>> 
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>> FreeSWITCH-dev at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> 
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-d
>> ev
>> http://www.freeswitch.org
>> 
> 
> --
> 
> <mod_bwe_debug.c>_______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
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From Giovanni.Visciano at italtel.it  Fri Oct 21 00:11:45 2011
From: Giovanni.Visciano at italtel.it (Visciano Giovanni)
Date: Thu, 20 Oct 2011 22:11:45 +0200
Subject: [Freeswitch-dev] Strange behaviour in SOFIA SIP
References: <AF4EB29748E178458A0067661112724B04EC6B1B@BESONE.corp.dom>
Message-ID: <AF4EB29748E178458A0067661112724B02B48F17@BESONE.corp.dom>

Preconditions on sip profile
----------------------------
inbound-codec-prefs NOT set
outbound-codec-prefs NOT set
inbound-proxy-media=TRUE
inbound-bypass-media=FALSE
inbound-late-negotiation=TRUE + inherit_codec=TRUE

TEST scenario
-------------
SIP phone send INVITE with SDP to FS

INVITE
SDP[codelist=PCMA, PCMU, G722, G729] --> FS

FS has just the CORE_PC_MODULE [PCMA, PCMU, PASS-THR, VIDEO-PASS-THR, L16]

The "ep_codec_string" channel variable is loaded with [PCMA, PCMU]

EXPECTED
--------
In "proxy-media + late-negotiation" scenario: ep_codec_string=[PCMA,PCMU,G722,G729]

PROBLEM
-------
According to the documentation (http://wiki.freeswitch.org/wiki/Codec_negotiation):

"ep_codec_string: This variable is only available if late negotiation is enabled on the profile. It's a readable string containing all the codecs proposed by the calling endpoint."
and
"inherit_codec=true (only applicable when late-negotiation is enabled) will take the codec negotiated when the B leg answers and pass it to the A leg so it also uses the same codec (if the A leg supports it); otherwise, it will use whatever it can from its own list"

The variable ep_codec_string doesn't contain ALL the codecs proposed by the calling endpoint. It contains the intersection between the codecs proposed by the calling and the codecs loaded in FS.
In proxy-media / no-media configuration the called endpoint doesn't have the calling codec list so he can't negotiate using all the capabilities available.
(I'm working on a test endpoint module that should work in proxy and no-media configuration ... mod_sofia vs mod_sofia works because they look at the remote sdp variable)

Source reference - sofia_glue.c->sofia_glue_set_r_sdp_codec_string()


Regards,
Giovanni


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From giovanni_visciano at yahoo.it  Fri Oct 21 09:34:49 2011
From: giovanni_visciano at yahoo.it (Giovanni Visciano)
Date: Fri, 21 Oct 2011 06:34:49 +0100 (BST)
Subject: [Freeswitch-dev] Sofia and "ep_codec_string"
Message-ID: <1319175289.44155.YahooMailNeo@web25703.mail.ukl.yahoo.com>

Preconditions on sip profile
----------------------------
inbound-codec-prefs NOT set
outbound-codec-prefs NOT set
inbound-proxy-media=TRUE
inbound-bypass-media=FALSE
inbound-late-negotiation=TRUE + inherit_codec=TRUE
?
TEST scenario
-------------
SIP phone send INVITE with SDP to FS
?
INVITE
SDP[codelist=PCMA, PCMU, G722, G729] --> FS
?
FS has just the CORE_PC_MODULE [PCMA, PCMU, PASS-THR,
VIDEO-PASS-THR, L16]
?
The "ep_codec_string" channel variable is
loaded with [PCMA, PCMU]
?
EXPECTED
--------
In "proxy-media + late-negotiation" scenario:
ep_codec_string=[PCMA,PCMU,G722,G729]
?
PROBLEM
-------
According to the documentation (http://wiki.freeswitch.org/wiki/Codec_negotiation):
?
"ep_codec_string: This variable is only available if
late negotiation is enabled on the profile. It's a readable string containing
all the codecs proposed by the calling endpoint."
and
"inherit_codec=true (only applicable when
late-negotiation is enabled) will take the codec negotiated when the B leg
answers and pass it to the A leg so it also uses the same codec (if the A leg
supports it); otherwise, it will use whatever it can from its own list"
?
The variable ep_codec_string doesn't contain ALL the
codecs proposed by the calling endpoint. It contains the intersection between
the codecs proposed by the calling and the codecs loaded in FS.
In proxy-media / no-media configuration the called
endpoint doesn't have the calling codec list so he can't negotiate using all
the capabilities available.
(I'm working on a test endpoint module that should work
in proxy and no-media configuration ... mod_sofia vs mod_sofia works because they
look at the remote sdp variable)
?
Source reference -
sofia_glue.c->sofia_glue_set_r_sdp_codec_string()
?
?
Regards,
Giovanni



From anthony.minessale at gmail.com  Fri Oct 21 19:06:34 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 21 Oct 2011 10:06:34 -0500
Subject: [Freeswitch-dev] Strange behaviour in SOFIA SIP
In-Reply-To: <AF4EB29748E178458A0067661112724B02B48F17@BESONE.corp.dom>
References: <AF4EB29748E178458A0067661112724B04EC6B1B@BESONE.corp.dom>
	<AF4EB29748E178458A0067661112724B02B48F17@BESONE.corp.dom>
Message-ID: <CAKbxfG-U3BU7mha9zeN-_cWpXjbhwMFBAkuva=rrg_dn8Nvz+w@mail.gmail.com>

any codec manipulation techniques and proxy media mode are not
compatible.  Proxy media mode means stay completely out of the codec
negotiation process.

On Thu, Oct 20, 2011 at 3:11 PM, Visciano Giovanni
<Giovanni.Visciano at italtel.it> wrote:
> Preconditions on sip profile
> ----------------------------
> inbound-codec-prefs NOT set
> outbound-codec-prefs NOT set
> inbound-proxy-media=TRUE
> inbound-bypass-media=FALSE
> inbound-late-negotiation=TRUE + inherit_codec=TRUE
>
> TEST scenario
> -------------
> SIP phone send INVITE with SDP to FS
>
> INVITE
> SDP[codelist=PCMA, PCMU, G722, G729] --> FS
>
> FS has just the CORE_PC_MODULE [PCMA, PCMU, PASS-THR, VIDEO-PASS-THR, L16]
>
> The "ep_codec_string" channel variable is loaded with [PCMA, PCMU]
>
> EXPECTED
> --------
> In "proxy-media + late-negotiation" scenario:
> ep_codec_string=[PCMA,PCMU,G722,G729]
>
> PROBLEM
> -------
> According to the documentation
> (http://wiki.freeswitch.org/wiki/Codec_negotiation):
>
> "ep_codec_string: This variable is only available if late negotiation is
> enabled on the profile. It's a readable string containing all the codecs
> proposed by the calling endpoint."
> and
> "inherit_codec=true (only applicable when late-negotiation is enabled) will
> take the codec negotiated when the B leg answers and pass it to the A leg so
> it also uses the same codec (if the A leg supports it); otherwise, it will
> use whatever it can from its own list"
>
> The variable ep_codec_string doesn't contain ALL the codecs proposed by the
> calling endpoint. It contains the intersection between the codecs proposed
> by the calling and the codecs loaded in FS.
> In proxy-media / no-media configuration the called endpoint doesn't have the
> calling codec list so he can't negotiate using all the capabilities
> available.
> (I'm working on a test endpoint module that should work in proxy and
> no-media configuration ... mod_sofia vs mod_sofia works because they look at
> the remote sdp variable)
>
> Source reference - sofia_glue.c->sofia_glue_set_r_sdp_codec_string()
>
>
> Regards,
> Giovanni
>
>
>
> Internet Email Confidentiality Footer
>
> ********************************************************************************************************************************************
>
> La presente comunicazione, con le informazioni in essa contenute e ogni
> documento o file allegato, e' rivolta unicamente alla/e persona/e cui e'
> indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete
> i destinatari/autorizzati siete avvisati che qualsiasi azione, copia,
> comunicazione, divulgazione o simili basate sul contenuto di tali
> informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P.,
> D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se
> avete ricevuto questa comunicazione per errore, vi preghiamo di darne
> immediata notizia al mittente e di distruggere il messaggio originale e ogni
> file allegato senza farne copia alcuna o riprodurne in alcun modo il
> contenuto.
>
> This e-mail and its attachments are intended for the addressee(s) only and
> are confidential and/or may contain legally privileged information. If you
> have received this message by mistake or are not one of the addressees
> above, you may take no action based on it, and you may not copy or show it
> to anyone; please reply to this e-mail and point out the error which has
> occurred.
> ********************************************************************************************************************************************
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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 Oct 21 19:07:22 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 21 Oct 2011 10:07:22 -0500
Subject: [Freeswitch-dev] trouble on how to remove the media bug from
 the session after the sessions are closed
In-Reply-To: <E8A95E8A-E508-4ED1-8CFB-D195DC27C8AB@avgs.ca>
References: <314ebfaf9a5bf1bbd700ce3ba8a7920b@gw.ind.rwth-aachen.de>
	<E8A95E8A-E508-4ED1-8CFB-D195DC27C8AB@avgs.ca>
Message-ID: <CAKbxfG-Hiip7Djtk_674vSJcQFVnVwn=S=50+HbRZziBV4DbhQ@mail.gmail.com>

ha, i missed that one! I knew it was something like that =p


On Thu, Oct 20, 2011 at 10:39 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
> Hi,
>
> On line 188, you call switch_core_session_get_partner. That function internally calls switch_core_session_locate(), you have to rwunlock it.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 2011-10-20, at 12:09 PM, Han Xin wrote:
>
>> I tried a lot in the past days but still got no progress. So I upload my code here to ask for help again.
>>
>> Thanks anyway!
>>
>> ----------------original message-----------------
>> From: "Anthony Minessale" anthony.minessale at gmail.com
>> To: freeswitch-dev at lists.freeswitch.org
>> Date: Mon, 17 Oct 2011 09:39:26 -0500
>> -------------------------------------------------
>>
>>
>>> I have no more guesses. You need to review your custom code and find
>>> out where you read locked your session.
>>>
>>>
>>> On Mon, Oct 17, 2011 at 2:38 AM, Han Xin xin at ind.rwth-aachen.de wrote:
>>>> no, no use of ivr_originate function.
>>>> I constructed a helper struct which include the remote session, could this
>>>> cause the session releasing problem?
>>>>
>>>> ----------------urspr?ngliche Nachricht-----------------
>>>> Von: "Anthony Minessale" anthony.minessale at gmail.com
>>>> An: freeswitch-dev at lists.freeswitch.org
>>>> Datum: Thu, 13 Oct 2011 15:55:35 -0500
>>>> -------------------------------------------------
>>>>
>>>>
>>>>> how about switch_ivr_originate() ?
>>>>>
>>>>> On Thu, Oct 13, 2011 at 2:19 AM, Han Xin xin at ind.rwth-aachen.de wrote:
>>>>>> yes I use a custom mod and the latest git on Oct 10 but I did not use the
>>>>>> function
>>>>>> "switch_core_session_locate".
>>>>>>
>>>>>> ----------------urspr?ngliche Nachricht-----------------
>>>>>> Von: "Anthony Minessale" anthony.minessale at gmail.com
>>>>>> An: freeswitch-dev at lists.freeswitch.org
>>>>>> Datum: Wed, 12 Oct 2011 17:06:41 -0500
>>>>>> -------------------------------------------------
>>>>>>
>>>>>>
>>>>>>> looks like you have some custom mods maybe you used
>>>>>>> switch_core_session_locate and did not rwunlock it?
>>>>>>> are you using latest GIT ?
>>>>>>>
>>>>>>>
>>>>>>> On Mon, Oct 10, 2011 at 11:03 AM, Han Xin xin at ind.rwth-aachen.de
>>>>>>> wrote:
>>>>>>>> yes, I use the SWITCH_ABC_TYPE_CLOSE type and I've found the
>>>>>>>> media
>>>>>>>> bugs are
>>>>>>>> automatically removed with the DEBUG log infomation. Now the
>>>>>>>> problem
>>>>>>>> is the
>>>>>>>> session ending.
>>>>>>>>
>>>>>>>> Detailed situation description: I added two media bugs to both
>>>>>>>> local
>>>>>>>> session
>>>>>>>> and remote session of one call. When I ended the call, the media
>>>>>>>> bugs are
>>>>>>>> removed
>>>>>>>> first, and then the session should be ended. But only the local
>>>>>>>> session
>>>>>>>> is ended
>>>>>>>> correctly.
>>>>>>>> When I press F3 and F4 to show current channels and calls, the
>>>>>>>> remote
>>>>>>>> channel
>>>>>>>> and call info is till there. But when I try to dump uuid to get
>>>>>>>> channel
>>>>>>>> infomation
>>>>>>>> with the cmd "uuid_dump remote_session_uuid", it says there
>>>>>>>> is no
>>>>>>>> such
>>>>>>>> channel!
>>>>>>>>
>>>>>>>> here is some related log info:
>>>>>>>>
>>>>>>>> 2011-10-10 17:36:41.428463 [ERR] mod_bwe.c:345
>>>>>>>> 28b3a542-aecd-4b9b-9f55-3c2674a25b5c: Remove local
>>>>>>>> variable
>>>>>>>> store!
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_ivr_bridge.c:1332
>>>>>>>> sofia/internal/sip:1001 at 137.226.198.252 :5060 skip
>>>>>>>> receive
>>>>>>>> message
>>>>>>>> [UNBRIDGE] (channel is hungup already)
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_ivr_bridge.c:1335
>>>>>>>> sofia/internal/1000 at asterisk skip receive message
>>>>>>>> [UNBRIDGE]
>>>>>>>> (channel is
>>>>>>>> hungup already)
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_media_bug.c:480
>>>>>>>> Removing BUG from
>>>>>>>> sofia/internal/sip:1001 at 137.226.198.252
>>>>>>>> :5060
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:602
>>>>>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State
>>>>>>>> HANGUP
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:454
>>>>>>>> sofia/internal/sip:1001 at 137.226.198.252 :5060
>>>>>>>> Overriding SIP
>>>>>>>> cause 480
>>>>>>>> with 200 from the other leg
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:460
>>>>>>>> Channel
>>>>>>>> sofia/internal/sip:1001 at 137.226.198.252 :5060 hanging
>>>>>>>> up,
>>>>>>>> cause:
>>>>>>>> NORMAL_CLEARING
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:503
>>>>>>>> Sending BYE
>>>>>>>> to
>>>>>>>> sofia/internal/sip:1001 at 137.226.198.252 :5060
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_session.c:2235
>>>>>>>> sofia/internal/1000 at asterisk skip receive message
>>>>>>>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:417
>>>>>>>> (sofia/internal/1000 at asterisk ) State EXECUTE going to
>>>>>>>> sleep
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:362
>>>>>>>> (sofia/internal/1000 at asterisk ) Running State Change
>>>>>>>> CS_HANGUP
>>>>>>>> 2011-10-10 17:36:41.428463 [ERR] mod_bwe.c:345
>>>>>>>> 713d5f0c-4341-47ec-b0bf-4b14bd6b4726: Remove local
>>>>>>>> variable
>>>>>>>> store!
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_media_bug.c:480
>>>>>>>> Removing BUG from sofia/internal/1000 at asterisk
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:602
>>>>>>>> (sofia/internal/1000 at asterisk ) State HANGUP
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:454
>>>>>>>> sofia/internal/1000 at asterisk Overriding SIP cause 480 with
>>>>>>>> 200
>>>>>>>> from the
>>>>>>>> other leg
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:460
>>>>>>>> Channel
>>>>>>>> sofia/internal/1000 at asterisk hanging up, cause:
>>>>>>>> NORMAL_CLEARING
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:47
>>>>>>>> sofia/internal/1000 at asterisk Standard HANGUP, cause:
>>>>>>>> NORMAL_CLEARING
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:47
>>>>>>>> sofia/internal/sip:1001 at 137.226.198.252 :5060 Standard
>>>>>>>> HANGUP,
>>>>>>>> cause:
>>>>>>>> NORMAL_CLEARING
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:602
>>>>>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State
>>>>>>>> HANGUP
>>>>>>>> going to
>>>>>>>> sleep
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:393
>>>>>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State
>>>>>>>> Change
>>>>>>>> CS_HANGUP
>>>>>>>> -> CS_REPORTING
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_session.c:1177
>>>>>>>> Send
>>>>>>>> signal sofia/internal/sip:1001 at 137.226.198.252 :5060
>>>>>>>> [BREAK]
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:362
>>>>>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) Running
>>>>>>>> State
>>>>>>>> Change
>>>>>>>> CS_REPORTING
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:662
>>>>>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State
>>>>>>>> REPORTING
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:79
>>>>>>>> sofia/internal/sip:1001 at 137.226.198.252 :5060 Standard
>>>>>>>> REPORTING,
>>>>>>>> cause: NORMAL_CLEARING
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:662
>>>>>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State
>>>>>>>> REPORTING going
>>>>>>>> to sleep
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:387
>>>>>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) State
>>>>>>>> Change
>>>>>>>> CS_REPORTING -> CS_DESTROY
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_session.c:1177
>>>>>>>> Send
>>>>>>>> signal sofia/internal/sip:1001 at 137.226.198.252 :5060
>>>>>>>> [BREAK]
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_session.c:1351
>>>>>>>> Session
>>>>>>>> 8 (sofia/internal/sip:1001 at 137.226.198.252 :5060)
>>>>>>>> Locked,
>>>>>>>> Waiting on
>>>>>>>> external entities
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:602
>>>>>>>> (sofia/internal/1000 at asterisk ) State HANGUP going to
>>>>>>>> sleep
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:393
>>>>>>>> (sofia/internal/1000 at asterisk ) State Change CS_HANGUP ->
>>>>>>>> CS_REPORTING
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_session.c:1177
>>>>>>>> Send
>>>>>>>> signal sofia/internal/1000 at asterisk [BREAK]
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:362
>>>>>>>> (sofia/internal/1000 at asterisk ) Running State Change
>>>>>>>> CS_REPORTING
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:662
>>>>>>>> (sofia/internal/1000 at asterisk ) State REPORTING
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:79
>>>>>>>> sofia/internal/1000 at asterisk Standard REPORTING, cause:
>>>>>>>> NORMAL_CLEARING
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:662
>>>>>>>> (sofia/internal/1000 at asterisk ) State REPORTING going to
>>>>>>>> sleep
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:387
>>>>>>>> (sofia/internal/1000 at asterisk ) State Change CS_REPORTING
>>>>>>>> ->
>>>>>>>> CS_DESTROY
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_session.c:1177
>>>>>>>> Send
>>>>>>>> signal sofia/internal/1000 at asterisk [BREAK]
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_session.c:1351
>>>>>>>> Session
>>>>>>>> 7 (sofia/internal/1000 at asterisk ) Locked, Waiting on
>>>>>>>> external
>>>>>>>> entities
>>>>>>>> 2011-10-10 17:36:41.428463 [NOTICE]
>>>>>>>> switch_core_session.c:1369
>>>>>>>> Session 7 (sofia/internal/1000 at asterisk ) Ended
>>>>>>>> 2011-10-10 17:36:41.428463 [NOTICE]
>>>>>>>> switch_core_session.c:1371
>>>>>>>> Close
>>>>>>>> Channel sofia/internal/1000 at asterisk [CS_DESTROY]
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:491
>>>>>>>> (sofia/internal/1000 at asterisk ) Callstate Change HANGUP ->
>>>>>>>> DOWN
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:494
>>>>>>>> (sofia/internal/1000 at asterisk ) Running State Change
>>>>>>>> CS_DESTROY
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:504
>>>>>>>> (sofia/internal/1000 at asterisk ) State DESTROY
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG] mod_sofia.c:365
>>>>>>>> sofia/internal/1000 at asterisk SOFIA DESTROY
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:86
>>>>>>>> sofia/internal/1000 at asterisk Standard DESTROY
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:504
>>>>>>>> (sofia/internal/1000 at asterisk ) State DESTROY going to
>>>>>>>> sleep
>>>>>>>>
>>>>>>>> Could anyone give me some hints why the remote session is not
>>>>>>>> ended
>>>>>>>> automatically?
>>>>>>>>
>>>>>>>>
>>>>>>>> ----------------urspr?ngliche
>>>>>>>> Nachricht-----------------
>>>>>>>> Von: "Anthony Minessale" anthony.minessale at gmail.com
>>>>>>>> An: freeswitch-dev at lists.freeswitch.org
>>>>>>>> Datum: Mon, 10 Oct 2011 10:17:36 -0500
>>>>>>>> -------------------------------------------------
>>>>>>>>
>>>>>>>>
>>>>>>>>> it calls the callback with type set to
>>>>>>>>> SWITCH_ABC_TYPE_CLOSE
>>>>>>>>>
>>>>>>>>> On Wed, Oct 5, 2011 at 7:37 PM, Vu Quang Hoa
>>>>>>>>> vutamhoan at gmail.com
>>>>>>>>> wrote:
>>>>>>>>>> As I tested - FS automatically removes bug when call
>>>>>>>>>> ends.
>>>>>>>>>> I'm trying to figure out if it does the same with
>>>>>>>>>> allocated
>>>>>>>>>> resources
>>>>>>>>>> in
>>>>>>>>>> add_bug routine.
>>>>>>>>>>
>>>>>>>>>> On Tue, Oct 4, 2011 at 7:03 PM, Han Xin
>>>>>>>>>> xin at ind.rwth-aachen.de
>>>>>>>>>> wrote:
>>>>>>>>>>>
>>>>>>>>>>> Hello everyone!
>>>>>>>>>>>
>>>>>>>>>>> I am new to Freeswitch and try to add media bugs such
>>>>>>>>>>> as
>>>>>>>>>>> digital
>>>>>>>>>>> signal
>>>>>>>>>>> processing to the read stream and write stream. But I
>>>>>>>>>>> have
>>>>>>>>>>> some
>>>>>>>>>>> trouble on
>>>>>>>>>>> how to remove the media bug from the session after
>>>>>>>>>>> the
>>>>>>>>>>> sessions are
>>>>>>>>>>> closed.
>>>>>>>>>>> I tried to look into other existed modules such as
>>>>>>>>>>> mod_soundtouch
>>>>>>>>>>> and
>>>>>>>>>>> mod_dptools to find a solution, the former
>>>>>>>>>>> mod_soundtouch
>>>>>>>>>>> put
>>>>>>>>>>> the
>>>>>>>>>>> function
>>>>>>>>>>> "switch_core_media_bug_remove()" in the
>>>>>>>>>>> beginning of
>>>>>>>>>>> application
>>>>>>>>>>> run
>>>>>>>>>>> function. The latter mod use a single app called
>>>>>>>>>>> "remove
>>>>>>>>>>> bugs" and
>>>>>>>>>>> put
>>>>>>>>>>> this
>>>>>>>>>>> app in the dial plan to remove the bugs.
>>>>>>>>>>>
>>>>>>>>>>> The question is where should I put this
>>>>>>>>>>> "switch_core_media_bug_remove()"
>>>>>>>>>>> function after the sessions are closed.
>>>>>>>>>>> Is there a way to automatically remove a media bug
>>>>>>>>>>> when a call
>>>>>>>>>>> is
>>>>>>>>>>> ended?
>>>>>>>>>>> Or it is only possible to use a single app to remove the
>>>>>>>>>>> media
>>>>>>>>>>> bugs as
>>>>>>>>>>> in
>>>>>>>>>>> the mod_dptools?
>>>>>>>>>>>
>>>>>>>>>>> Thank you all in advance!
>>>>>>>>>>>
>>>>>>>>>>> --
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>>
>>>>>>>>>>> FreeSWITCH-dev mailing list
>>>>>>>>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswi
>>>>>>>>>>> tch-
>>>>>>>>>>> dev
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opti
>>>>>>>>>>> ons/
>>>>>>>>>>> free
>>>>>>>>>>> swit
>>>>>>>>>>> ch-dev
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>
>>>>>>>>>> FreeSWITCH-dev mailing list
>>>>>>>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitc
>>>>>>>>>> h-de
>>>>>>>>>> v
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/option
>>>>>>>>>> s/fr
>>>>>>>>>> eesw
>>>>>>>>>> itch
>>>>>>>>>> -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
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>
>>>>>>>>> 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/
>>>>>>>>> free
>>>>>>>>> swit
>>>>>>>>> ch-d
>>>>>>>>> ev
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>
>>>>>>>> --
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>
>>>>>>>> FreeSWITCH-dev mailing list
>>>>>>>> FreeSWITCH-dev at lists.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-de
>>>>>>>> v
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/fr
>>>>>>>> eesw
>>>>>>>> itch
>>>>>>>> -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
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>
>>>>>>> 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/free
>>>>>>> swit
>>>>>>> ch-d
>>>>>>> ev
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>> --
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> 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/freesw
>>>>>> itch
>>>>>> -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
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> 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/freeswit
>>>>> ch-d
>>>>> ev
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>> --
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> 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
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> 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-d
>>> ev
>>> http://www.freeswitch.org
>>>
>>
>> --
>>
>> <mod_bwe_debug.c>_______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> 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
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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 Oct 21 20:03:07 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 21 Oct 2011 11:03:07 -0500
Subject: [Freeswitch-dev] Calling Python Application from Chatplan
 (mod_sms help)
In-Reply-To: <CACyT-3=2gk=7hXAca-QAm27s-1f0eefJrz=v3xrfbhmZUS2ksw@mail.gmail.com>
References: <CACyT-3mxBZQax2A94MHbOSXm6mwgAm_uD4fJAxqOi_iBRKrLag@mail.gmail.com>
	<CACyT-3mWkP6mg8OS342K617SP6dsnhGX+9yM9vy2SsBYW_0qsw@mail.gmail.com>
	<CAKbxfG89z+EWZn0+Zw44zJgbq52MQLxgACZFJ1_1-7zTDGnSHg@mail.gmail.com>
	<CACyT-3nDD-SePsvj=6Sypxa+ZzC0S0rkrtL-HRqYTHHJd7JWBQ@mail.gmail.com>
	<CACyT-3mYZAA66dx7yLX5rdDhs-1Hcm+GbuW6HiypaUPXBcHSwA@mail.gmail.com>
	<CAKbxfG8PibyP-ymp63miAD0GjO5ytJZ4xi88c7BeWHwRm=UpzQ@mail.gmail.com>
	<CACyT-3=2gk=7hXAca-QAm27s-1f0eefJrz=v3xrfbhmZUS2ksw@mail.gmail.com>
Message-ID: <CAKbxfG_0qv1ystkxPaqXAxrUUr3kB6dT5crq4fp3t8QM4nuaLw@mail.gmail.com>

there wasn't but I added it to latest rev.

getGlobalVariable and setGlobalVariable

used in same context as consoleLog


On Thu, Oct 20, 2011 at 9:25 PM, Kurtis Heimerl
<kheimerl at cs.berkeley.edu> wrote:
> Sorry about that, you're right. For some reason, python wasn't
> building with my main install. Got that remedied and everything seems
> to be working! Thanks!
>
> One quick question, is there any way to access session variables
> (e.g., ${domain}, or anything from vars.xml) from the "message" object
> that freeswitch hands a chat application?
>
> On Thu, Oct 20, 2011 at 2:37 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> from the code:
>>
>> SWITCH_ADD_CHAT_APP(chat_app_interface, "python", "execute a python
>> script", "execute a python script", python_chat_function, "<script>",
>> SCAF_NONE);
>>
>> you must not have the version loaded that implemented it.
>>
>>
>> On Thu, Oct 20, 2011 at 3:40 PM, Kurtis Heimerl
>> <kheimerl at cs.berkeley.edu> wrote:
>>> Still not working:
>>>
>>> ? ?<extension name="openbts">
>>> ? ? ?<condition field="to_user" expression="^smsc$">
>>> ? ? ? ?<action application="set" data="api_result=${log(${to})}"/>
>>> ? ? ? ?<action application="python" data="OpenBTS_Parse_SMS"/>
>>> ? ? ?</condition>
>>> ? ?</extension>
>>>
>>> 2011-10-20 13:39:49.534714 [INFO] mod_sms.c:300 Processing text
>>> message IMSI641104278340879->smsc in context default
>>> 2011-10-20 13:39:49.534714 [ERR] switch_loadable_module.c:705 Invalid
>>> chat application interface [python]!
>>>
>>> On Wed, Oct 19, 2011 at 9:57 AM, Kurtis Heimerl
>>> <kheimerl at cs.berkeley.edu> wrote:
>>>> Great, will do!
>>>>
>>>> On Wed, Oct 19, 2011 at 8:23 AM, Anthony Minessale
>>>> <anthony.minessale at gmail.com> wrote:
>>>>> try latest
>>>>>
>>>>> On Tue, Oct 18, 2011 at 7:00 PM, Kurtis Heimerl
>>>>> <kheimerl at cs.berkeley.edu> wrote:
>>>>>> Oh, and I can confirm that only Lua works from the chatplan.
>>>>>>
>>>>>> On Tue, Oct 18, 2011 at 4:12 PM, Kurtis Heimerl
>>>>>> <kheimerl at cs.berkeley.edu> wrote:
>>>>>>> I'm pushing this to FreeSwitch-dev as mod_sms is still under active
>>>>>>> development.
>>>>>>>
>>>>>>> Any way to call a Python Application from the chatplan?
>>>>>>>
>>>>>>> This works:
>>>>>>>
>>>>>>> <action application="set" data="test=${python(OpenBTS_Parse_SMS)}"/>
>>>>>>>
>>>>>>> But I can't monkey with the session variables with the fs_api call.
>>>>>>>
>>>>>>> Instead, I want to do this:
>>>>>>>
>>>>>>> <action application="python" data="OpenBTS_Parse_SMS"/>
>>>>>>>
>>>>>>> and this fails in the following way:
>>>>>>>
>>>>>>> 2011-10-18 16:12:02.776298 [ERR] switch_loadable_module.c:705 Invalid
>>>>>>> chat application interface [python]!
>>>>>>>
>>>>>>> The mod_sms page seems to indicate that lua works just fine from the
>>>>>>> chatplan... is this just not implemented yet? Can I do that?
>>>>>>>
>>>>>>> Thanks!
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> 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
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> 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
>>>>>
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> 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
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> 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
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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 kheimerl at cs.berkeley.edu  Sat Oct 22 04:20:20 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Fri, 21 Oct 2011 17:20:20 -0700
Subject: [Freeswitch-dev] Call-ID formatting
Message-ID: <CACyT-3=VbQO28bQ5xyZLMQZXYLNmNn1rtAt1Ej-5EN2yzizv_w@mail.gmail.com>

According to the sip standard, call-ids should have a "host" trailer
(e.g., f81d4fae-7dec-11d0-a765-00a0c91e6bf6 at foo.bar.com)

Reference: http://www.ietf.org/rfc/rfc2543.txt Section 6.12

FreeSwitch doesn't do this, omitting the "@foo.bar.com" portion of the
above. Is there any reason why this is the case? This seems too large
of an issue to be a bug or oversight.

Here's an example from a FS MESSAGE:

Call-ID: 000b3694-76e4-122f-55b3-5cff350d9de5


From kris at kriskinc.com  Sat Oct 22 04:33:28 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Fri, 21 Oct 2011 20:33:28 -0400
Subject: [Freeswitch-dev] Call-ID formatting
In-Reply-To: <CACyT-3=VbQO28bQ5xyZLMQZXYLNmNn1rtAt1Ej-5EN2yzizv_w@mail.gmail.com>
References: <CACyT-3=VbQO28bQ5xyZLMQZXYLNmNn1rtAt1Ej-5EN2yzizv_w@mail.gmail.com>
Message-ID: <CAKDfjgdDmkuh6X3AVCLRXzzXgdXtAkQO9w-k6gYvz6nv1WXqnA@mail.gmail.com>

RFC3261 defines SIP 2.0.  2543 is deprecated.  3261 section 8.1.1.4 says:

"Use of cryptographically random identifiers (RFC 1750 [12]) in the
   generation of Call-IDs is RECOMMENDED.  Implementations MAY use the
   form "localid at host".  Call-IDs are case-sensitive and are simply
   compared byte-by-byte."

On Fri, Oct 21, 2011 at 8:20 PM, Kurtis Heimerl
<kheimerl at cs.berkeley.edu> wrote:
> According to the sip standard, call-ids should have a "host" trailer
> (e.g., f81d4fae-7dec-11d0-a765-00a0c91e6bf6 at foo.bar.com)
>
> Reference: http://www.ietf.org/rfc/rfc2543.txt Section 6.12
>
> FreeSwitch doesn't do this, omitting the "@foo.bar.com" portion of the
> above. Is there any reason why this is the case? This seems too large
> of an issue to be a bug or oversight.
>
> Here's an example from a FS MESSAGE:
>
> Call-ID: 000b3694-76e4-122f-55b3-5cff350d9de5
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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
>



-- 
Kristian Kielhofner


From kheimerl at cs.berkeley.edu  Sat Oct 22 04:36:07 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Fri, 21 Oct 2011 17:36:07 -0700
Subject: [Freeswitch-dev] Call-ID formatting
In-Reply-To: <CAKDfjgdDmkuh6X3AVCLRXzzXgdXtAkQO9w-k6gYvz6nv1WXqnA@mail.gmail.com>
References: <CACyT-3=VbQO28bQ5xyZLMQZXYLNmNn1rtAt1Ej-5EN2yzizv_w@mail.gmail.com>
	<CAKDfjgdDmkuh6X3AVCLRXzzXgdXtAkQO9w-k6gYvz6nv1WXqnA@mail.gmail.com>
Message-ID: <CACyT-3nK+LDfC-RmD2fz4O44uJ21r5r6Th7KEAYr-owoyW0psg@mail.gmail.com>

Got it! Thanks!

On Fri, Oct 21, 2011 at 5:33 PM, Kristian Kielhofner <kris at kriskinc.com> wrote:
> RFC3261 defines SIP 2.0. ?2543 is deprecated. ?3261 section 8.1.1.4 says:
>
> "Use of cryptographically random identifiers (RFC 1750 [12]) in the
> ? generation of Call-IDs is RECOMMENDED. ?Implementations MAY use the
> ? form "localid at host". ?Call-IDs are case-sensitive and are simply
> ? compared byte-by-byte."
>
> On Fri, Oct 21, 2011 at 8:20 PM, Kurtis Heimerl
> <kheimerl at cs.berkeley.edu> wrote:
>> According to the sip standard, call-ids should have a "host" trailer
>> (e.g., f81d4fae-7dec-11d0-a765-00a0c91e6bf6 at foo.bar.com)
>>
>> Reference: http://www.ietf.org/rfc/rfc2543.txt Section 6.12
>>
>> FreeSwitch doesn't do this, omitting the "@foo.bar.com" portion of the
>> above. Is there any reason why this is the case? This seems too large
>> of an issue to be a bug or oversight.
>>
>> Here's an example from a FS MESSAGE:
>>
>> Call-ID: 000b3694-76e4-122f-55b3-5cff350d9de5
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> 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
>>
>
>
>
> --
> Kristian Kielhofner
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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 kris at kriskinc.com  Sat Oct 22 04:45:28 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Fri, 21 Oct 2011 20:45:28 -0400
Subject: [Freeswitch-dev] Call-ID formatting
In-Reply-To: <CAKDfjgdDmkuh6X3AVCLRXzzXgdXtAkQO9w-k6gYvz6nv1WXqnA@mail.gmail.com>
References: <CACyT-3=VbQO28bQ5xyZLMQZXYLNmNn1rtAt1Ej-5EN2yzizv_w@mail.gmail.com>
	<CAKDfjgdDmkuh6X3AVCLRXzzXgdXtAkQO9w-k6gYvz6nv1WXqnA@mail.gmail.com>
Message-ID: <CAKDfjgdJH8i27jktzq=81478bhABti=L47KxRpEx6Yr08Bxq6Q@mail.gmail.com>

I just realized this is ambiguous...  Technically 2543 defined "SIP
2.0" but "SIP 2.0" behavior today is generally considered to be 3261
compliant, not 2543 compliant although you still run into 2543
behavior in various ways and it's not necessarily "wrong".

On Fri, Oct 21, 2011 at 8:33 PM, Kristian Kielhofner <kris at kriskinc.com> wrote:
> RFC3261 defines SIP 2.0. ?2543 is deprecated. ?3261 section 8.1.1.4 says:
>
> "Use of cryptographically random identifiers (RFC 1750 [12]) in the
> ? generation of Call-IDs is RECOMMENDED. ?Implementations MAY use the
> ? form "localid at host". ?Call-IDs are case-sensitive and are simply
> ? compared byte-by-byte."
>
> On Fri, Oct 21, 2011 at 8:20 PM, Kurtis Heimerl
> <kheimerl at cs.berkeley.edu> wrote:
>> According to the sip standard, call-ids should have a "host" trailer
>> (e.g., f81d4fae-7dec-11d0-a765-00a0c91e6bf6 at foo.bar.com)
>>
>> Reference: http://www.ietf.org/rfc/rfc2543.txt Section 6.12
>>
>> FreeSwitch doesn't do this, omitting the "@foo.bar.com" portion of the
>> above. Is there any reason why this is the case? This seems too large
>> of an issue to be a bug or oversight.
>>
>> Here's an example from a FS MESSAGE:
>>
>> Call-ID: 000b3694-76e4-122f-55b3-5cff350d9de5
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> 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
>>
>
>
>
> --
> Kristian Kielhofner
>



-- 
Kristian Kielhofner


From kheimerl at cs.berkeley.edu  Sat Oct 22 06:22:55 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Fri, 21 Oct 2011 19:22:55 -0700
Subject: [Freeswitch-dev] Call-ID formatting
In-Reply-To: <CAKDfjgdJH8i27jktzq=81478bhABti=L47KxRpEx6Yr08Bxq6Q@mail.gmail.com>
References: <CACyT-3=VbQO28bQ5xyZLMQZXYLNmNn1rtAt1Ej-5EN2yzizv_w@mail.gmail.com>
	<CAKDfjgdDmkuh6X3AVCLRXzzXgdXtAkQO9w-k6gYvz6nv1WXqnA@mail.gmail.com>
	<CAKDfjgdJH8i27jktzq=81478bhABti=L47KxRpEx6Yr08Bxq6Q@mail.gmail.com>
Message-ID: <CACyT-3mQWHj+vbj7KhJSAaHHvgn=WJmTQtdUST7T17mZ_pCNAQ@mail.gmail.com>

I'm working on a sip client that segfaulted without the host, so
that's clearly incorrect. Thanks for the direction.

On Fri, Oct 21, 2011 at 5:45 PM, Kristian Kielhofner <kris at kriskinc.com> wrote:
> I just realized this is ambiguous... ?Technically 2543 defined "SIP
> 2.0" but "SIP 2.0" behavior today is generally considered to be 3261
> compliant, not 2543 compliant although you still run into 2543
> behavior in various ways and it's not necessarily "wrong".
>
> On Fri, Oct 21, 2011 at 8:33 PM, Kristian Kielhofner <kris at kriskinc.com> wrote:
>> RFC3261 defines SIP 2.0. ?2543 is deprecated. ?3261 section 8.1.1.4 says:
>>
>> "Use of cryptographically random identifiers (RFC 1750 [12]) in the
>> ? generation of Call-IDs is RECOMMENDED. ?Implementations MAY use the
>> ? form "localid at host". ?Call-IDs are case-sensitive and are simply
>> ? compared byte-by-byte."
>>
>> On Fri, Oct 21, 2011 at 8:20 PM, Kurtis Heimerl
>> <kheimerl at cs.berkeley.edu> wrote:
>>> According to the sip standard, call-ids should have a "host" trailer
>>> (e.g., f81d4fae-7dec-11d0-a765-00a0c91e6bf6 at foo.bar.com)
>>>
>>> Reference: http://www.ietf.org/rfc/rfc2543.txt Section 6.12
>>>
>>> FreeSwitch doesn't do this, omitting the "@foo.bar.com" portion of the
>>> above. Is there any reason why this is the case? This seems too large
>>> of an issue to be a bug or oversight.
>>>
>>> Here's an example from a FS MESSAGE:
>>>
>>> Call-ID: 000b3694-76e4-122f-55b3-5cff350d9de5
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> 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
>>>
>>
>>
>>
>> --
>> Kristian Kielhofner
>>
>
>
>
> --
> Kristian Kielhofner
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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  Sat Oct 22 06:40:49 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 21 Oct 2011 21:40:49 -0500
Subject: [Freeswitch-dev] Call-ID formatting
In-Reply-To: <CACyT-3mQWHj+vbj7KhJSAaHHvgn=WJmTQtdUST7T17mZ_pCNAQ@mail.gmail.com>
References: <CACyT-3=VbQO28bQ5xyZLMQZXYLNmNn1rtAt1Ej-5EN2yzizv_w@mail.gmail.com>
	<CAKDfjgdDmkuh6X3AVCLRXzzXgdXtAkQO9w-k6gYvz6nv1WXqnA@mail.gmail.com>
	<CAKDfjgdJH8i27jktzq=81478bhABti=L47KxRpEx6Yr08Bxq6Q@mail.gmail.com>
	<CACyT-3mQWHj+vbj7KhJSAaHHvgn=WJmTQtdUST7T17mZ_pCNAQ@mail.gmail.com>
Message-ID: <CAKbxfG_8B7twwx91Rs4XwS1XB_7PutH3v3CzZJOV612r7f63=w@mail.gmail.com>

does that have anything to do with this bug about message and contact?


On Fri, Oct 21, 2011 at 9:22 PM, Kurtis Heimerl
<kheimerl at cs.berkeley.edu> wrote:
> I'm working on a sip client that segfaulted without the host, so
> that's clearly incorrect. Thanks for the direction.
>
> On Fri, Oct 21, 2011 at 5:45 PM, Kristian Kielhofner <kris at kriskinc.com> wrote:
>> I just realized this is ambiguous... ?Technically 2543 defined "SIP
>> 2.0" but "SIP 2.0" behavior today is generally considered to be 3261
>> compliant, not 2543 compliant although you still run into 2543
>> behavior in various ways and it's not necessarily "wrong".
>>
>> On Fri, Oct 21, 2011 at 8:33 PM, Kristian Kielhofner <kris at kriskinc.com> wrote:
>>> RFC3261 defines SIP 2.0. ?2543 is deprecated. ?3261 section 8.1.1.4 says:
>>>
>>> "Use of cryptographically random identifiers (RFC 1750 [12]) in the
>>> ? generation of Call-IDs is RECOMMENDED. ?Implementations MAY use the
>>> ? form "localid at host". ?Call-IDs are case-sensitive and are simply
>>> ? compared byte-by-byte."
>>>
>>> On Fri, Oct 21, 2011 at 8:20 PM, Kurtis Heimerl
>>> <kheimerl at cs.berkeley.edu> wrote:
>>>> According to the sip standard, call-ids should have a "host" trailer
>>>> (e.g., f81d4fae-7dec-11d0-a765-00a0c91e6bf6 at foo.bar.com)
>>>>
>>>> Reference: http://www.ietf.org/rfc/rfc2543.txt Section 6.12
>>>>
>>>> FreeSwitch doesn't do this, omitting the "@foo.bar.com" portion of the
>>>> above. Is there any reason why this is the case? This seems too large
>>>> of an issue to be a bug or oversight.
>>>>
>>>> Here's an example from a FS MESSAGE:
>>>>
>>>> Call-ID: 000b3694-76e4-122f-55b3-5cff350d9de5
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> 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
>>>>
>>>
>>>
>>>
>>> --
>>> Kristian Kielhofner
>>>
>>
>>
>>
>> --
>> Kristian Kielhofner
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> 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
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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 kheimerl at cs.berkeley.edu  Sat Oct 22 06:47:07 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Fri, 21 Oct 2011 19:47:07 -0700
Subject: [Freeswitch-dev] Call-ID formatting
In-Reply-To: <CAKbxfG_8B7twwx91Rs4XwS1XB_7PutH3v3CzZJOV612r7f63=w@mail.gmail.com>
References: <CACyT-3=VbQO28bQ5xyZLMQZXYLNmNn1rtAt1Ej-5EN2yzizv_w@mail.gmail.com>
	<CAKDfjgdDmkuh6X3AVCLRXzzXgdXtAkQO9w-k6gYvz6nv1WXqnA@mail.gmail.com>
	<CAKDfjgdJH8i27jktzq=81478bhABti=L47KxRpEx6Yr08Bxq6Q@mail.gmail.com>
	<CACyT-3mQWHj+vbj7KhJSAaHHvgn=WJmTQtdUST7T17mZ_pCNAQ@mail.gmail.com>
	<CAKbxfG_8B7twwx91Rs4XwS1XB_7PutH3v3CzZJOV612r7f63=w@mail.gmail.com>
Message-ID: <CACyT-3mkiN2gTDiVFBZZ-ut61YtHa-j6Cm6KGtKRA7bF74s1hA@mail.gmail.com>

No, that's a separate issue, though it's totally possible that there's
ANOTHER rfc that I've missed reading for that one.

Simple story: RFC 3428 (http://www.ietf.org/rfc/rfc3428.txt) says that
a SIP MESSAGE should not have a contact header, and it does.


On Fri, Oct 21, 2011 at 7:40 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> does that have anything to do with this bug about message and contact?
>
>
> On Fri, Oct 21, 2011 at 9:22 PM, Kurtis Heimerl
> <kheimerl at cs.berkeley.edu> wrote:
>> I'm working on a sip client that segfaulted without the host, so
>> that's clearly incorrect. Thanks for the direction.
>>
>> On Fri, Oct 21, 2011 at 5:45 PM, Kristian Kielhofner <kris at kriskinc.com> wrote:
>>> I just realized this is ambiguous... ?Technically 2543 defined "SIP
>>> 2.0" but "SIP 2.0" behavior today is generally considered to be 3261
>>> compliant, not 2543 compliant although you still run into 2543
>>> behavior in various ways and it's not necessarily "wrong".
>>>
>>> On Fri, Oct 21, 2011 at 8:33 PM, Kristian Kielhofner <kris at kriskinc.com> wrote:
>>>> RFC3261 defines SIP 2.0. ?2543 is deprecated. ?3261 section 8.1.1.4 says:
>>>>
>>>> "Use of cryptographically random identifiers (RFC 1750 [12]) in the
>>>> ? generation of Call-IDs is RECOMMENDED. ?Implementations MAY use the
>>>> ? form "localid at host". ?Call-IDs are case-sensitive and are simply
>>>> ? compared byte-by-byte."
>>>>
>>>> On Fri, Oct 21, 2011 at 8:20 PM, Kurtis Heimerl
>>>> <kheimerl at cs.berkeley.edu> wrote:
>>>>> According to the sip standard, call-ids should have a "host" trailer
>>>>> (e.g., f81d4fae-7dec-11d0-a765-00a0c91e6bf6 at foo.bar.com)
>>>>>
>>>>> Reference: http://www.ietf.org/rfc/rfc2543.txt Section 6.12
>>>>>
>>>>> FreeSwitch doesn't do this, omitting the "@foo.bar.com" portion of the
>>>>> above. Is there any reason why this is the case? This seems too large
>>>>> of an issue to be a bug or oversight.
>>>>>
>>>>> Here's an example from a FS MESSAGE:
>>>>>
>>>>> Call-ID: 000b3694-76e4-122f-55b3-5cff350d9de5
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> 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
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Kristian Kielhofner
>>>>
>>>
>>>
>>>
>>> --
>>> Kristian Kielhofner
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> 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
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> 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
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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  Sat Oct 22 06:53:26 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 21 Oct 2011 21:53:26 -0500
Subject: [Freeswitch-dev] Call-ID formatting
In-Reply-To: <CACyT-3mkiN2gTDiVFBZZ-ut61YtHa-j6Cm6KGtKRA7bF74s1hA@mail.gmail.com>
References: <CACyT-3=VbQO28bQ5xyZLMQZXYLNmNn1rtAt1Ej-5EN2yzizv_w@mail.gmail.com>
	<CAKDfjgdDmkuh6X3AVCLRXzzXgdXtAkQO9w-k6gYvz6nv1WXqnA@mail.gmail.com>
	<CAKDfjgdJH8i27jktzq=81478bhABti=L47KxRpEx6Yr08Bxq6Q@mail.gmail.com>
	<CACyT-3mQWHj+vbj7KhJSAaHHvgn=WJmTQtdUST7T17mZ_pCNAQ@mail.gmail.com>
	<CAKbxfG_8B7twwx91Rs4XwS1XB_7PutH3v3CzZJOV612r7f63=w@mail.gmail.com>
	<CACyT-3mkiN2gTDiVFBZZ-ut61YtHa-j6Cm6KGtKRA7bF74s1hA@mail.gmail.com>
Message-ID: <CAKbxfG8Es0bqDNbChvtubPW037CqpU7BFM5j67tn0Nxg3rjd4Q@mail.gmail.com>

I was just curious why you found that?  Its been that way for years
and nothing has ever complained.
Did you just have it memorized or hit an interop issue or what?


On Fri, Oct 21, 2011 at 9:47 PM, Kurtis Heimerl
<kheimerl at cs.berkeley.edu> wrote:
> No, that's a separate issue, though it's totally possible that there's
> ANOTHER rfc that I've missed reading for that one.
>
> Simple story: RFC 3428 (http://www.ietf.org/rfc/rfc3428.txt) says that
> a SIP MESSAGE should not have a contact header, and it does.
>
>
> On Fri, Oct 21, 2011 at 7:40 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> does that have anything to do with this bug about message and contact?
>>
>>
>> On Fri, Oct 21, 2011 at 9:22 PM, Kurtis Heimerl
>> <kheimerl at cs.berkeley.edu> wrote:
>>> I'm working on a sip client that segfaulted without the host, so
>>> that's clearly incorrect. Thanks for the direction.
>>>
>>> On Fri, Oct 21, 2011 at 5:45 PM, Kristian Kielhofner <kris at kriskinc.com> wrote:
>>>> I just realized this is ambiguous... ?Technically 2543 defined "SIP
>>>> 2.0" but "SIP 2.0" behavior today is generally considered to be 3261
>>>> compliant, not 2543 compliant although you still run into 2543
>>>> behavior in various ways and it's not necessarily "wrong".
>>>>
>>>> On Fri, Oct 21, 2011 at 8:33 PM, Kristian Kielhofner <kris at kriskinc.com> wrote:
>>>>> RFC3261 defines SIP 2.0. ?2543 is deprecated. ?3261 section 8.1.1.4 says:
>>>>>
>>>>> "Use of cryptographically random identifiers (RFC 1750 [12]) in the
>>>>> ? generation of Call-IDs is RECOMMENDED. ?Implementations MAY use the
>>>>> ? form "localid at host". ?Call-IDs are case-sensitive and are simply
>>>>> ? compared byte-by-byte."
>>>>>
>>>>> On Fri, Oct 21, 2011 at 8:20 PM, Kurtis Heimerl
>>>>> <kheimerl at cs.berkeley.edu> wrote:
>>>>>> According to the sip standard, call-ids should have a "host" trailer
>>>>>> (e.g., f81d4fae-7dec-11d0-a765-00a0c91e6bf6 at foo.bar.com)
>>>>>>
>>>>>> Reference: http://www.ietf.org/rfc/rfc2543.txt Section 6.12
>>>>>>
>>>>>> FreeSwitch doesn't do this, omitting the "@foo.bar.com" portion of the
>>>>>> above. Is there any reason why this is the case? This seems too large
>>>>>> of an issue to be a bug or oversight.
>>>>>>
>>>>>> Here's an example from a FS MESSAGE:
>>>>>>
>>>>>> Call-ID: 000b3694-76e4-122f-55b3-5cff350d9de5
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> 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
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Kristian Kielhofner
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Kristian Kielhofner
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> 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
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> 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
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> 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
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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 kheimerl at cs.berkeley.edu  Sat Oct 22 06:58:24 2011
From: kheimerl at cs.berkeley.edu (Kurtis Heimerl)
Date: Fri, 21 Oct 2011 19:58:24 -0700
Subject: [Freeswitch-dev] Call-ID formatting
In-Reply-To: <CAKbxfG8Es0bqDNbChvtubPW037CqpU7BFM5j67tn0Nxg3rjd4Q@mail.gmail.com>
References: <CACyT-3=VbQO28bQ5xyZLMQZXYLNmNn1rtAt1Ej-5EN2yzizv_w@mail.gmail.com>
	<CAKDfjgdDmkuh6X3AVCLRXzzXgdXtAkQO9w-k6gYvz6nv1WXqnA@mail.gmail.com>
	<CAKDfjgdJH8i27jktzq=81478bhABti=L47KxRpEx6Yr08Bxq6Q@mail.gmail.com>
	<CACyT-3mQWHj+vbj7KhJSAaHHvgn=WJmTQtdUST7T17mZ_pCNAQ@mail.gmail.com>
	<CAKbxfG_8B7twwx91Rs4XwS1XB_7PutH3v3CzZJOV612r7f63=w@mail.gmail.com>
	<CACyT-3mkiN2gTDiVFBZZ-ut61YtHa-j6Cm6KGtKRA7bF74s1hA@mail.gmail.com>
	<CAKbxfG8Es0bqDNbChvtubPW037CqpU7BFM5j67tn0Nxg3rjd4Q@mail.gmail.com>
Message-ID: <CACyT-3nRnGWpVdLsFb5jufzTewAR6VpejP6q=qXwOT=400VwNw@mail.gmail.com>

I've run into a particularly pedantic RFC 3428 store-and-forward server, yeah.

If it's a hard fix, I can push on them to fix that instead of FS, but
it'll be hard if the spec is on their side.

On Fri, Oct 21, 2011 at 7:53 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> I was just curious why you found that? ?Its been that way for years
> and nothing has ever complained.
> Did you just have it memorized or hit an interop issue or what?
>
>
> On Fri, Oct 21, 2011 at 9:47 PM, Kurtis Heimerl
> <kheimerl at cs.berkeley.edu> wrote:
>> No, that's a separate issue, though it's totally possible that there's
>> ANOTHER rfc that I've missed reading for that one.
>>
>> Simple story: RFC 3428 (http://www.ietf.org/rfc/rfc3428.txt) says that
>> a SIP MESSAGE should not have a contact header, and it does.
>>
>>
>> On Fri, Oct 21, 2011 at 7:40 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>>> does that have anything to do with this bug about message and contact?
>>>
>>>
>>> On Fri, Oct 21, 2011 at 9:22 PM, Kurtis Heimerl
>>> <kheimerl at cs.berkeley.edu> wrote:
>>>> I'm working on a sip client that segfaulted without the host, so
>>>> that's clearly incorrect. Thanks for the direction.
>>>>
>>>> On Fri, Oct 21, 2011 at 5:45 PM, Kristian Kielhofner <kris at kriskinc.com> wrote:
>>>>> I just realized this is ambiguous... ?Technically 2543 defined "SIP
>>>>> 2.0" but "SIP 2.0" behavior today is generally considered to be 3261
>>>>> compliant, not 2543 compliant although you still run into 2543
>>>>> behavior in various ways and it's not necessarily "wrong".
>>>>>
>>>>> On Fri, Oct 21, 2011 at 8:33 PM, Kristian Kielhofner <kris at kriskinc.com> wrote:
>>>>>> RFC3261 defines SIP 2.0. ?2543 is deprecated. ?3261 section 8.1.1.4 says:
>>>>>>
>>>>>> "Use of cryptographically random identifiers (RFC 1750 [12]) in the
>>>>>> ? generation of Call-IDs is RECOMMENDED. ?Implementations MAY use the
>>>>>> ? form "localid at host". ?Call-IDs are case-sensitive and are simply
>>>>>> ? compared byte-by-byte."
>>>>>>
>>>>>> On Fri, Oct 21, 2011 at 8:20 PM, Kurtis Heimerl
>>>>>> <kheimerl at cs.berkeley.edu> wrote:
>>>>>>> According to the sip standard, call-ids should have a "host" trailer
>>>>>>> (e.g., f81d4fae-7dec-11d0-a765-00a0c91e6bf6 at foo.bar.com)
>>>>>>>
>>>>>>> Reference: http://www.ietf.org/rfc/rfc2543.txt Section 6.12
>>>>>>>
>>>>>>> FreeSwitch doesn't do this, omitting the "@foo.bar.com" portion of the
>>>>>>> above. Is there any reason why this is the case? This seems too large
>>>>>>> of an issue to be a bug or oversight.
>>>>>>>
>>>>>>> Here's an example from a FS MESSAGE:
>>>>>>>
>>>>>>> Call-ID: 000b3694-76e4-122f-55b3-5cff350d9de5
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>
>>>>>>> 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
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Kristian Kielhofner
>>>>>>
>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Kristian Kielhofner
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> 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
>>>>>
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> 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
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> 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
>>>
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> 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
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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  Sat Oct 22 07:03:51 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 21 Oct 2011 22:03:51 -0500
Subject: [Freeswitch-dev] Call-ID formatting
In-Reply-To: <CACyT-3nRnGWpVdLsFb5jufzTewAR6VpejP6q=qXwOT=400VwNw@mail.gmail.com>
References: <CACyT-3=VbQO28bQ5xyZLMQZXYLNmNn1rtAt1Ej-5EN2yzizv_w@mail.gmail.com>
	<CAKDfjgdDmkuh6X3AVCLRXzzXgdXtAkQO9w-k6gYvz6nv1WXqnA@mail.gmail.com>
	<CAKDfjgdJH8i27jktzq=81478bhABti=L47KxRpEx6Yr08Bxq6Q@mail.gmail.com>
	<CACyT-3mQWHj+vbj7KhJSAaHHvgn=WJmTQtdUST7T17mZ_pCNAQ@mail.gmail.com>
	<CAKbxfG_8B7twwx91Rs4XwS1XB_7PutH3v3CzZJOV612r7f63=w@mail.gmail.com>
	<CACyT-3mkiN2gTDiVFBZZ-ut61YtHa-j6Cm6KGtKRA7bF74s1hA@mail.gmail.com>
	<CAKbxfG8Es0bqDNbChvtubPW037CqpU7BFM5j67tn0Nxg3rjd4Q@mail.gmail.com>
	<CACyT-3nRnGWpVdLsFb5jufzTewAR6VpejP6q=qXwOT=400VwNw@mail.gmail.com>
Message-ID: <CAKbxfG_gvOg4bJKN+S=5zD-4bMVxRJMMt2qnYFDLnWM-JtPvng@mail.gmail.com>

nah, it's trivial, I just hope it won't break all those non-pedantic
things out there.... =D


On Fri, Oct 21, 2011 at 9:58 PM, Kurtis Heimerl
<kheimerl at cs.berkeley.edu> wrote:
> I've run into a particularly pedantic RFC 3428 store-and-forward server, yeah.
>
> If it's a hard fix, I can push on them to fix that instead of FS, but
> it'll be hard if the spec is on their side.
>
> On Fri, Oct 21, 2011 at 7:53 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> I was just curious why you found that? ?Its been that way for years
>> and nothing has ever complained.
>> Did you just have it memorized or hit an interop issue or what?
>>
>>
>> On Fri, Oct 21, 2011 at 9:47 PM, Kurtis Heimerl
>> <kheimerl at cs.berkeley.edu> wrote:
>>> No, that's a separate issue, though it's totally possible that there's
>>> ANOTHER rfc that I've missed reading for that one.
>>>
>>> Simple story: RFC 3428 (http://www.ietf.org/rfc/rfc3428.txt) says that
>>> a SIP MESSAGE should not have a contact header, and it does.
>>>
>>>
>>> On Fri, Oct 21, 2011 at 7:40 PM, Anthony Minessale
>>> <anthony.minessale at gmail.com> wrote:
>>>> does that have anything to do with this bug about message and contact?
>>>>
>>>>
>>>> On Fri, Oct 21, 2011 at 9:22 PM, Kurtis Heimerl
>>>> <kheimerl at cs.berkeley.edu> wrote:
>>>>> I'm working on a sip client that segfaulted without the host, so
>>>>> that's clearly incorrect. Thanks for the direction.
>>>>>
>>>>> On Fri, Oct 21, 2011 at 5:45 PM, Kristian Kielhofner <kris at kriskinc.com> wrote:
>>>>>> I just realized this is ambiguous... ?Technically 2543 defined "SIP
>>>>>> 2.0" but "SIP 2.0" behavior today is generally considered to be 3261
>>>>>> compliant, not 2543 compliant although you still run into 2543
>>>>>> behavior in various ways and it's not necessarily "wrong".
>>>>>>
>>>>>> On Fri, Oct 21, 2011 at 8:33 PM, Kristian Kielhofner <kris at kriskinc.com> wrote:
>>>>>>> RFC3261 defines SIP 2.0. ?2543 is deprecated. ?3261 section 8.1.1.4 says:
>>>>>>>
>>>>>>> "Use of cryptographically random identifiers (RFC 1750 [12]) in the
>>>>>>> ? generation of Call-IDs is RECOMMENDED. ?Implementations MAY use the
>>>>>>> ? form "localid at host". ?Call-IDs are case-sensitive and are simply
>>>>>>> ? compared byte-by-byte."
>>>>>>>
>>>>>>> On Fri, Oct 21, 2011 at 8:20 PM, Kurtis Heimerl
>>>>>>> <kheimerl at cs.berkeley.edu> wrote:
>>>>>>>> According to the sip standard, call-ids should have a "host" trailer
>>>>>>>> (e.g., f81d4fae-7dec-11d0-a765-00a0c91e6bf6 at foo.bar.com)
>>>>>>>>
>>>>>>>> Reference: http://www.ietf.org/rfc/rfc2543.txt Section 6.12
>>>>>>>>
>>>>>>>> FreeSwitch doesn't do this, omitting the "@foo.bar.com" portion of the
>>>>>>>> above. Is there any reason why this is the case? This seems too large
>>>>>>>> of an issue to be a bug or oversight.
>>>>>>>>
>>>>>>>> Here's an example from a FS MESSAGE:
>>>>>>>>
>>>>>>>> Call-ID: 000b3694-76e4-122f-55b3-5cff350d9de5
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>
>>>>>>>> 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
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> Kristian Kielhofner
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Kristian Kielhofner
>>>>>>
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>
>>>>>> 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
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>
>>>>> 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
>>>>
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>
>>>> 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
>>>>
>>>
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>>
>>> 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
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> 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
>>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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 giovanni_visciano at yahoo.it  Fri Oct 21 11:45:01 2011
From: giovanni_visciano at yahoo.it (Giovanni Visciano)
Date: Fri, 21 Oct 2011 08:45:01 +0100 (BST)
Subject: [Freeswitch-dev] Sofia and "ep_codec_string"
In-Reply-To: <A2167AE7633A5C4DAE56EEBD1428ECC0F36D6D@BESONE.corp.dom>
References: <A2167AE7633A5C4DAE56EEBD1428ECC0F36D6D@BESONE.corp.dom>
Message-ID: <1319183101.19288.YahooMailNeo@web25708.mail.ukl.yahoo.com>

Sorry for the double posting. I sent this question also from my office email giovanni.visciano at italtel.it.
Please ignore this one.

Thanks.
Giovanni

-----Original Message-----
From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Giovanni Visciano
Sent: Friday, October 21, 2011 7:35 AM
To: freeswitch-dev at lists.freeswitch.org
Subject: [Freeswitch-dev] Sofia and "ep_codec_string"

Preconditions on sip profile
----------------------------
inbound-codec-prefs NOT set
outbound-codec-prefs NOT set
inbound-proxy-media=TRUE
inbound-bypass-media=FALSE
inbound-late-negotiation=TRUE + inherit_codec=TRUE

TEST scenario
-------------
SIP phone send INVITE with SDP to FS

INVITE
SDP[codelist=PCMA, PCMU, G722, G729] --> FS

FS has just the CORE_PC_MODULE [PCMA, PCMU, PASS-THR,
VIDEO-PASS-THR, L16]

The "ep_codec_string" channel variable is
loaded with [PCMA, PCMU]

EXPECTED
--------
In "proxy-media + late-negotiation" scenario:
ep_codec_string=[PCMA,PCMU,G722,G729]

PROBLEM
-------
According to the documentation (http://wiki.freeswitch.org/wiki/Codec_negotiation):

"ep_codec_string: This variable is only available if
late negotiation is enabled on the profile. It's a readable string containing
all the codecs proposed by the calling endpoint."
and
"inherit_codec=true (only applicable when
late-negotiation is enabled) will take the codec negotiated when the B leg
answers and pass it to the A leg so it also uses the same codec (if the A leg
supports it); otherwise, it will use whatever it can from its own list"

The variable ep_codec_string doesn't contain ALL the
codecs proposed by the calling endpoint. It contains the intersection between
the codecs proposed by the calling and the codecs loaded in FS.
In proxy-media / no-media configuration the called
endpoint doesn't have the calling codec list so he can't negotiate using all
the capabilities available.
(I'm working on a test endpoint module that should work
in proxy and no-media configuration ... mod_sofia vs mod_sofia works because they
look at the remote sdp variable)

Source reference -
sofia_glue.c->sofia_glue_set_r_sdp_codec_string()


Regards,
Giovanni


From Giovanni.Visciano at italtel.it  Fri Oct 21 19:36:11 2011
From: Giovanni.Visciano at italtel.it (Visciano Giovanni)
Date: Fri, 21 Oct 2011 17:36:11 +0200
Subject: [Freeswitch-dev] Strange behaviour in SOFIA SIP
References: <AF4EB29748E178458A0067661112724B04EC6B1B@BESONE.corp.dom><AF4EB29748E178458A0067661112724B02B48F17@BESONE.corp.dom>
	<CAKbxfG-U3BU7mha9zeN-_cWpXjbhwMFBAkuva=rrg_dn8Nvz+w@mail.gmail.com>
Message-ID: <AF4EB29748E178458A0067661112724B02B48F19@BESONE.corp.dom>

Hello Anthony.
Maybe I want not so clear, let me explain.

I do not want to apply any codec manipulation, I'm working on a prototype endpoint module that has "proxy media capabilty" and wants to interoperate with other endpoints.
What I saw is that in a sofia vs sofia call (proxy mode+late negotiation) they exchange through FS calling/called endpoint capability looking at SDP session carried by channel variables, so internally they use SDP O/A scheme.
I thought that "ep_codec_string"->SWITCH_ORIGINATOR_CODEC_VARIABLE could be used to exchange capability without using a protocol specific format (SDP).
If this is not the case, what's the right way to negotiate capability between endpoints in proxy mode?

Than you.
Giovanni

PS: I did not find any existing endpoint except sofia that support this. Am I right?

-----Original Message-----
From: freeswitch-dev-bounces at lists.freeswitch.org on behalf of Anthony Minessale
Sent: Fri 10/21/2011 5:06 PM
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] Strange behaviour in SOFIA SIP
 
any codec manipulation techniques and proxy media mode are not
compatible.  Proxy media mode means stay completely out of the codec
negotiation process.

On Thu, Oct 20, 2011 at 3:11 PM, Visciano Giovanni
<Giovanni.Visciano at italtel.it> wrote:
> Preconditions on sip profile
> ----------------------------
> inbound-codec-prefs NOT set
> outbound-codec-prefs NOT set
> inbound-proxy-media=TRUE
> inbound-bypass-media=FALSE
> inbound-late-negotiation=TRUE + inherit_codec=TRUE
>
> TEST scenario
> -------------
> SIP phone send INVITE with SDP to FS
>
> INVITE
> SDP[codelist=PCMA, PCMU, G722, G729] --> FS
>
> FS has just the CORE_PC_MODULE [PCMA, PCMU, PASS-THR, VIDEO-PASS-THR, L16]
>
> The "ep_codec_string" channel variable is loaded with [PCMA, PCMU]
>
> EXPECTED
> --------
> In "proxy-media + late-negotiation" scenario:
> ep_codec_string=[PCMA,PCMU,G722,G729]
>
> PROBLEM
> -------
> According to the documentation
> (http://wiki.freeswitch.org/wiki/Codec_negotiation):
>
> "ep_codec_string: This variable is only available if late negotiation is
> enabled on the profile. It's a readable string containing all the codecs
> proposed by the calling endpoint."
> and
> "inherit_codec=true (only applicable when late-negotiation is enabled) will
> take the codec negotiated when the B leg answers and pass it to the A leg so
> it also uses the same codec (if the A leg supports it); otherwise, it will
> use whatever it can from its own list"
>
> The variable ep_codec_string doesn't contain ALL the codecs proposed by the
> calling endpoint. It contains the intersection between the codecs proposed
> by the calling and the codecs loaded in FS.
> In proxy-media / no-media configuration the called endpoint doesn't have the
> calling codec list so he can't negotiate using all the capabilities
> available.
> (I'm working on a test endpoint module that should work in proxy and
> no-media configuration ... mod_sofia vs mod_sofia works because they look at
> the remote sdp variable)
>
> Source reference - sofia_glue.c->sofia_glue_set_r_sdp_codec_string()
>
>
> Regards,
> Giovanni
>
>
>
> Internet Email Confidentiality Footer
>
> ********************************************************************************************************************************************
>
> La presente comunicazione, con le informazioni in essa contenute e ogni
> documento o file allegato, e' rivolta unicamente alla/e persona/e cui e'
> indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete
> i destinatari/autorizzati siete avvisati che qualsiasi azione, copia,
> comunicazione, divulgazione o simili basate sul contenuto di tali
> informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P.,
> D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se
> avete ricevuto questa comunicazione per errore, vi preghiamo di darne
> immediata notizia al mittente e di distruggere il messaggio originale e ogni
> file allegato senza farne copia alcuna o riprodurne in alcun modo il
> contenuto.
>
> This e-mail and its attachments are intended for the addressee(s) only and
> are confidential and/or may contain legally privileged information. If you
> have received this message by mistake or are not one of the addressees
> above, you may take no action based on it, and you may not copy or show it
> to anyone; please reply to this e-mail and point out the error which has
> occurred.
> ********************************************************************************************************************************************
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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

_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

FreeSWITCH-dev mailing list
FreeSWITCH-dev at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
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From anthony.minessale at gmail.com  Sat Oct 22 18:48:05 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 22 Oct 2011 09:48:05 -0500
Subject: [Freeswitch-dev] Strange behaviour in SOFIA SIP
In-Reply-To: <AF4EB29748E178458A0067661112724B02B48F19@BESONE.corp.dom>
References: <AF4EB29748E178458A0067661112724B04EC6B1B@BESONE.corp.dom>
	<AF4EB29748E178458A0067661112724B02B48F17@BESONE.corp.dom>
	<CAKbxfG-U3BU7mha9zeN-_cWpXjbhwMFBAkuva=rrg_dn8Nvz+w@mail.gmail.com>
	<AF4EB29748E178458A0067661112724B02B48F19@BESONE.corp.dom>
Message-ID: <CAKbxfG_Y4Q50kFryKEonNAJ9f1pGUcTwE6haH1GjcVAkUYwsQw@mail.gmail.com>

that's what I was trying to explain.
PROXY mode does not let you manipulate anything, it's pure
pass-through negotiation between the endpoints.

All the stuff with ep_codec_string are relevant only when you are not
trying to use proxy mode.
This is a common mistake people make assuming not using proxy mode
means somehow doing more work but it's not the case, any time both
legs of a call use the same codec it's equally efficient to what you
end up with in proxy mode.  proxy mode simply means take the inbound
SDP and offer it AS IS to the far end and vice-versa and use a fake
codec in the middle that is agnostic.



On Fri, Oct 21, 2011 at 10:36 AM, Visciano Giovanni
<Giovanni.Visciano at italtel.it> wrote:
> Hello Anthony.
> Maybe I want not so clear, let me explain.
>
> I do not want to apply any codec manipulation, I'm working on a prototype endpoint module that has "proxy media capabilty" and wants to interoperate with other endpoints.
> What I saw is that in a sofia vs sofia call (proxy mode+late negotiation) they exchange through FS calling/called endpoint capability looking at SDP session carried by channel variables, so internally they use SDP O/A scheme.
> I thought that "ep_codec_string"->SWITCH_ORIGINATOR_CODEC_VARIABLE could be used to exchange capability without using a protocol specific format (SDP).
> If this is not the case, what's the right way to negotiate capability between endpoints in proxy mode?
>
> Than you.
> Giovanni
>
> PS: I did not find any existing endpoint except sofia that support this. Am I right?
>
> -----Original Message-----
> From: freeswitch-dev-bounces at lists.freeswitch.org on behalf of Anthony Minessale
> Sent: Fri 10/21/2011 5:06 PM
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] Strange behaviour in SOFIA SIP
>
> any codec manipulation techniques and proxy media mode are not
> compatible. ?Proxy media mode means stay completely out of the codec
> negotiation process.
>
> On Thu, Oct 20, 2011 at 3:11 PM, Visciano Giovanni
> <Giovanni.Visciano at italtel.it> wrote:
>> Preconditions on sip profile
>> ----------------------------
>> inbound-codec-prefs NOT set
>> outbound-codec-prefs NOT set
>> inbound-proxy-media=TRUE
>> inbound-bypass-media=FALSE
>> inbound-late-negotiation=TRUE + inherit_codec=TRUE
>>
>> TEST scenario
>> -------------
>> SIP phone send INVITE with SDP to FS
>>
>> INVITE
>> SDP[codelist=PCMA, PCMU, G722, G729] --> FS
>>
>> FS has just the CORE_PC_MODULE [PCMA, PCMU, PASS-THR, VIDEO-PASS-THR, L16]
>>
>> The "ep_codec_string" channel variable is loaded with [PCMA, PCMU]
>>
>> EXPECTED
>> --------
>> In "proxy-media + late-negotiation" scenario:
>> ep_codec_string=[PCMA,PCMU,G722,G729]
>>
>> PROBLEM
>> -------
>> According to the documentation
>> (http://wiki.freeswitch.org/wiki/Codec_negotiation):
>>
>> "ep_codec_string: This variable is only available if late negotiation is
>> enabled on the profile. It's a readable string containing all the codecs
>> proposed by the calling endpoint."
>> and
>> "inherit_codec=true (only applicable when late-negotiation is enabled) will
>> take the codec negotiated when the B leg answers and pass it to the A leg so
>> it also uses the same codec (if the A leg supports it); otherwise, it will
>> use whatever it can from its own list"
>>
>> The variable ep_codec_string doesn't contain ALL the codecs proposed by the
>> calling endpoint. It contains the intersection between the codecs proposed
>> by the calling and the codecs loaded in FS.
>> In proxy-media / no-media configuration the called endpoint doesn't have the
>> calling codec list so he can't negotiate using all the capabilities
>> available.
>> (I'm working on a test endpoint module that should work in proxy and
>> no-media configuration ... mod_sofia vs mod_sofia works because they look at
>> the remote sdp variable)
>>
>> Source reference - sofia_glue.c->sofia_glue_set_r_sdp_codec_string()
>>
>>
>> Regards,
>> Giovanni
>>
>>
>>
>> Internet Email Confidentiality Footer
>>
>> ********************************************************************************************************************************************
>>
>> La presente comunicazione, con le informazioni in essa contenute e ogni
>> documento o file allegato, e' rivolta unicamente alla/e persona/e cui e'
>> indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete
>> i destinatari/autorizzati siete avvisati che qualsiasi azione, copia,
>> comunicazione, divulgazione o simili basate sul contenuto di tali
>> informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P.,
>> D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se
>> avete ricevuto questa comunicazione per errore, vi preghiamo di darne
>> immediata notizia al mittente e di distruggere il messaggio originale e ogni
>> file allegato senza farne copia alcuna o riprodurne in alcun modo il
>> contenuto.
>>
>> This e-mail and its attachments are intended for the addressee(s) only and
>> are confidential and/or may contain legally privileged information. If you
>> have received this message by mistake or are not one of the addressees
>> above, you may take no action based on it, and you may not copy or show it
>> to anyone; please reply to this e-mail and point out the error which has
>> occurred.
>> ********************************************************************************************************************************************
>>
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>>
>> 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
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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 blaisounou at gmail.com  Sat Oct 22 18:59:56 2011
From: blaisounou at gmail.com (marco blaisou)
Date: Sat, 22 Oct 2011 16:59:56 +0200
Subject: [Freeswitch-dev] == ISSUE building FreeSwitch on Windows
Message-ID: <CADLgD8_sGEb7VQbzkX8yZGcz2kg9Sn=2uuxNfHXxC9_OonsHfQ@mail.gmail.com>

Hi guys,
For the few days I've been unsuccesfull in building FreeSwith on Windows 7

I did follow the Wiki line per line, but some thing is really going wrong:
> apparently the files in /libs directory are not loaded from the net (ie
the util.vbs does seem to load and unpack the libs)
error is : *freeswitch\libs\win32\util.vbs(184, 2) msxml3.dll: Le syst?me ne
trouve pas la ressource sp?cifi?e (ie system doesn't find specified
resource)*

> then obviously the compiler starts complaining about missing stuff such
as:
*c:\users\jc\desktop\voip\freeswitch\freeswitch\src\mod\endpoints\mod_sofia\mod_sofia.h(47)
: fatal error C1083: Cannot open include file: 'switch_version.h': No such
file or directory*

To summarize:
I loaded the FreeSwitch git repository with TortoiseGit

My config is :
Windows 7/64bits
VSC2008 sp1
Windows SDK 7.1

My firewall is desactivated as suggested

I did follow the recommendation to clear some files:
*del libs\apr\include\apr.h
del libs\js\config.h
del libs\js\src\jsautocfg.h
del libs\js\nsprpub\pr\include\prcpucfg.h
del libs\iksemel\include\config.h
del libs\xmlrpc-c\xmlrpc_config.h
del libs\libsndfile\src\sfconfig.h
del libs\sofia-sip\libsofia-sip-ua\su\sofia-sip\su_configure.h
del libs\js\nsprpub\pr\include\prcpucfg.h

cd libs\win32\libsndfile
rename config.h sfconfig.h*

Could a guru provide some hints on how to solve this???
Thanks
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From Giovanni.Visciano at italtel.it  Sat Oct 22 22:30:06 2011
From: Giovanni.Visciano at italtel.it (Visciano Giovanni)
Date: Sat, 22 Oct 2011 20:30:06 +0200
Subject: [Freeswitch-dev] Strange behaviour in SOFIA SIP
References: <AF4EB29748E178458A0067661112724B04EC6B1B@BESONE.corp.dom><AF4EB29748E178458A0067661112724B02B48F17@BESONE.corp.dom><CAKbxfG-U3BU7mha9zeN-_cWpXjbhwMFBAkuva=rrg_dn8Nvz+w@mail.gmail.com><AF4EB29748E178458A0067661112724B02B48F19@BESONE.corp.dom>
	<CAKbxfG_Y4Q50kFryKEonNAJ9f1pGUcTwE6haH1GjcVAkUYwsQw@mail.gmail.com>
Message-ID: <AF4EB29748E178458A0067661112724B02B48F1A@BESONE.corp.dom>

Now I understand.
If both legs use the same codec, the codec ("fake") will be used in a pass-through way,
so no additional transcoding is applied, as in PROXY PASS-THROUGH condition.

So let me reason in a default media scenario (no proxy-media).

Now, as I explained in the first mail, sofia build "ep_codec_string" (SWITCH_
ORIGINATOR_CODEC_VARIABLE) as the intersection between the codec list received
in SDP offer and the codec loaded in the core.
So if the offered SDP codec list is NOT a subset of the loaded core codecs the other
legs will miss some codecs proposed by the calling endpoint.
That said, if I want to be sure every codecs my endpoint support won't be filtered I
have to register a "fake" pass-through codec for each one.

... you did not answer to my last question :).
I haven't found any other endpoint module the can interwork in PROXY mode with
sofia (even h323 doesn't support this). My question: is there a standard/
recommended way to exchange capability between endpoint of different nature in FS
proxy mode? Maybe using SDP o/a as done in sofia sip?

Thank you again for you help.
Giovanni


-----Original Message-----
From: freeswitch-dev-bounces at lists.freeswitch.org on behalf of Anthony Minessale
Sent: Sat 10/22/2011 4:48 PM
To: freeswitch-dev at lists.freeswitch.org
Subject: Re: [Freeswitch-dev] Strange behaviour in SOFIA SIP
 
that's what I was trying to explain.
PROXY mode does not let you manipulate anything, it's pure
pass-through negotiation between the endpoints.

All the stuff with ep_codec_string are relevant only when you are not
trying to use proxy mode.
This is a common mistake people make assuming not using proxy mode
means somehow doing more work but it's not the case, any time both
legs of a call use the same codec it's equally efficient to what you
end up with in proxy mode.  proxy mode simply means take the inbound
SDP and offer it AS IS to the far end and vice-versa and use a fake
codec in the middle that is agnostic.



On Fri, Oct 21, 2011 at 10:36 AM, Visciano Giovanni
<Giovanni.Visciano at italtel.it> wrote:
> Hello Anthony.
> Maybe I want not so clear, let me explain.
>
> I do not want to apply any codec manipulation, I'm working on a prototype endpoint module that has "proxy media capabilty" and wants to interoperate with other endpoints.
> What I saw is that in a sofia vs sofia call (proxy mode+late negotiation) they exchange through FS calling/called endpoint capability looking at SDP session carried by channel variables, so internally they use SDP O/A scheme.
> I thought that "ep_codec_string"->SWITCH_ORIGINATOR_CODEC_VARIABLE could be used to exchange capability without using a protocol specific format (SDP).
> If this is not the case, what's the right way to negotiate capability between endpoints in proxy mode?
>
> Than you.
> Giovanni
>
> PS: I did not find any existing endpoint except sofia that support this. Am I right?
>
> -----Original Message-----
> From: freeswitch-dev-bounces at lists.freeswitch.org on behalf of Anthony Minessale
> Sent: Fri 10/21/2011 5:06 PM
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] Strange behaviour in SOFIA SIP
>
> any codec manipulation techniques and proxy media mode are not
> compatible. ?Proxy media mode means stay completely out of the codec
> negotiation process.
>
> On Thu, Oct 20, 2011 at 3:11 PM, Visciano Giovanni
> <Giovanni.Visciano at italtel.it> wrote:
>> Preconditions on sip profile
>> ----------------------------
>> inbound-codec-prefs NOT set
>> outbound-codec-prefs NOT set
>> inbound-proxy-media=TRUE
>> inbound-bypass-media=FALSE
>> inbound-late-negotiation=TRUE + inherit_codec=TRUE
>>
>> TEST scenario
>> -------------
>> SIP phone send INVITE with SDP to FS
>>
>> INVITE
>> SDP[codelist=PCMA, PCMU, G722, G729] --> FS
>>
>> FS has just the CORE_PC_MODULE [PCMA, PCMU, PASS-THR, VIDEO-PASS-THR, L16]
>>
>> The "ep_codec_string" channel variable is loaded with [PCMA, PCMU]
>>
>> EXPECTED
>> --------
>> In "proxy-media + late-negotiation" scenario:
>> ep_codec_string=[PCMA,PCMU,G722,G729]
>>
>> PROBLEM
>> -------
>> According to the documentation
>> (http://wiki.freeswitch.org/wiki/Codec_negotiation):
>>
>> "ep_codec_string: This variable is only available if late negotiation is
>> enabled on the profile. It's a readable string containing all the codecs
>> proposed by the calling endpoint."
>> and
>> "inherit_codec=true (only applicable when late-negotiation is enabled) will
>> take the codec negotiated when the B leg answers and pass it to the A leg so
>> it also uses the same codec (if the A leg supports it); otherwise, it will
>> use whatever it can from its own list"
>>
>> The variable ep_codec_string doesn't contain ALL the codecs proposed by the
>> calling endpoint. It contains the intersection between the codecs proposed
>> by the calling and the codecs loaded in FS.
>> In proxy-media / no-media configuration the called endpoint doesn't have the
>> calling codec list so he can't negotiate using all the capabilities
>> available.
>> (I'm working on a test endpoint module that should work in proxy and
>> no-media configuration ... mod_sofia vs mod_sofia works because they look at
>> the remote sdp variable)
>>
>> Source reference - sofia_glue.c->sofia_glue_set_r_sdp_codec_string()
>>
>>
>> Regards,
>> Giovanni
>>
>>
>>
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>
> --
> Anthony Minessale II
>
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> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
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>
> _______________________________________________
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-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com
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_______________________________________________
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From peter.olsson at visionutveckling.se  Sun Oct 23 10:42:42 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sun, 23 Oct 2011 08:42:42 +0200
Subject: [Freeswitch-dev] == ISSUE building FreeSwitch on Windows
In-Reply-To: <CADLgD8_sGEb7VQbzkX8yZGcz2kg9Sn=2uuxNfHXxC9_OonsHfQ@mail.gmail.com>
References: <CADLgD8_sGEb7VQbzkX8yZGcz2kg9Sn=2uuxNfHXxC9_OonsHfQ@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF171@cooper>

I believe that you might be missing some system dll's on your machine.

msxml3.dll is something that should exist on your computer, and I think the download script is depending on it.

/Peter
________________________________________
Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [freeswitch-dev-bounces at lists.freeswitch.org] f&#246;r marco blaisou [blaisounou at gmail.com]
Skickat: den 22 oktober 2011 16:59
Till: freeswitch-dev at lists.freeswitch.org
?mne: [Freeswitch-dev] == ISSUE building FreeSwitch on Windows

Hi guys,
For the few days I've been unsuccesfull in building FreeSwith on Windows 7

I did follow the Wiki line per line, but some thing is really going wrong:
> apparently the files in /libs directory are not loaded from the net (ie the util.vbs does seem to load and unpack the libs)
error is : freeswitch\libs\win32\util.vbs(184, 2) msxml3.dll: Le syst?me ne trouve pas la ressource sp?cifi?e (ie system doesn't find specified resource)

> then obviously the compiler starts complaining about missing stuff such as:
c:\users\jc\desktop\voip\freeswitch\freeswitch\src\mod\endpoints\mod_sofia\mod_sofia.h(47) : fatal error C1083: Cannot open include file: 'switch_version.h': No such file or directory

To summarize:
I loaded the FreeSwitch git repository with TortoiseGit

My config is :
Windows 7/64bits
VSC2008 sp1
Windows SDK 7.1

My firewall is desactivated as suggested

I did follow the recommendation to clear some files:
del libs\apr\include\apr.h
del libs\js\config.h
del libs\js\src\jsautocfg.h
del libs\js\nsprpub\pr\include\prcpucfg.h
del libs\iksemel\include\config.h
del libs\xmlrpc-c\xmlrpc_config.h
del libs\libsndfile\src\sfconfig.h
del libs\sofia-sip\libsofia-sip-ua\su\sofia-sip\su_configure.h
del libs\js\nsprpub\pr\include\prcpucfg.h

cd libs\win32\libsndfile
rename config.h sfconfig.h

Could a guru provide some hints on how to solve this???
Thanks


!DSPAM:4ea3271432761137115228!


From juraj.fabo at gmail.com  Sun Oct 23 21:19:36 2011
From: juraj.fabo at gmail.com (Juraj Fabo)
Date: Sun, 23 Oct 2011 19:19:36 +0200
Subject: [Freeswitch-dev] MTU setting and application buffer size
Message-ID: <CAAqVCySU2mwpLzja4fhHN4YC-b4DJzvFj1GeWxixOev3F5NH9A@mail.gmail.com>

Hi

thank you for your response.

Here is my code used for testing the latency http://pastebin.com/mwHrRDeK.
It is based on and built as freeswitch-git/libs/freetdm/src/testapp.c
For the measurement I used a104d card with interconnected spans 1&2.
App loads binary file "sample_binin" which was previously created by dumping
the E1 channel when there was active call.
This dump is used as a source of voice.
App initiates a call on span1. App handles it as incoming call on span2.

After call is established, measurement loop is entered,measurement is in
single thread world, handling the spans 1 & 2 in "serial".
App feeds prebuffered voice to outgoing call on span1 and in paralell makes
a dump of written data to binary file "owbinout".
At the same time, data read from the span1 are dumped to binary file
"orbinout".
Then span2 is handled, data read from the active channel are just written
back, here the loopback is created.
In paralell data are dumped to file "awbinout"

So then results are being calculated by comparing the content of :
owbinout against awbinout = one way delay
owbinout against orbinout = roundtrip delay

Personally I use Audacity to import the binary voice, decode it and pair in
stereotracks so I can measure the delay.

With this approach I got the results posted earlier.

With best regards

Juraj Fabo



> ---------- Forwarded message ----------
> From: Moises Silva <moises.silva at gmail.com>
> To: freeswitch-dev at lists.freeswitch.org
> Date: Mon, 17 Oct 2011 12:22:58 -0400
> Subject: Re: [Freeswitch-dev] MTU setting and application buffer size
> On Tue, Oct 4, 2011 at 8:56 AM, Juraj Fabo <juraj.fabo at gmail.com> wrote:
>
>> > Please, have a look at the results with various parameters used:
>> > mtu:80 txqueue_size=1  rxqueue_size=1   one-direction delay=60ms
>> > round-trip delay=120ms
>> > mtu:80 txqueue_size=2  rxqueue_size=2   one-direction delay=80ms
>> > round-trip delay=160ms
>> > mtu:80 txqueue_size=10 rxqueue_size=10 one-direction delay=240ms
>> > round-trip delay=480ms
>> > mtu:40 txqueue_size=1  rxqueue_size=1   one-direction delay=40ms
>> > round-trip delay=80ms
>> > mtu:40 txqueue_size=2  rxqueue_size=2   one-direction delay=60ms
>> > round-trip delay=120ms
>> > mtu:40 txqueue_size=5  rxqueue_size=5   one-direction delay=120ms
>> > round-trip delay=240ms
>> > mtu:16 txqueue_size=1  rxqueue_size=1   one-direction delay=30ms
>> > round-trip delay=60ms
>> > mtu:8  txqueue_size=1   rxqueue_size=1   one-direction delay=25ms
>> > round-trip delay=50ms
>> > mtu:8  txqueue_size=10 rxqueue_size=10 one-direction delay=205ms
>> > round-trip delay=410ms
>> >
>>
>>
>> Updated test result with the codec_ms=10 is following:
>> mtu:16 txqueue_size=1  rxqueue_size=10   one-direction delay=20ms
>> round-trip delay=50ms
>> mtu:80 txqueue_size=1  rxqueue_size=10   one-direction delay=56ms
>> round-trip delay=110ms
>>
>> However, this is still not 10ms with the default MTU:80 and codec_ms:10 :(
>>
>> Anyway, is it a good idea to use so assymetric tx/rx queue sizes?
>>
>>
> In general I don't think is a good idea because 99% of the user population
> is using default symmetric queues, moving to obscure/rarely-tested
> configurations increases your likelihood of facing problems. Having said
> that, I don't see why it should not work :-)
>
> Can you share the code you're using to measure latency? may be setup a git
> repo in github with your code?
>
>  *Moises Silva
> **Software Engineer, Development Manager***
>
> msilva 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 x128
>
> f.   +1 905 474 9223
>
>
>
> **<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email+signatures>
>
> Products<http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>  | Solutions<http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>  | Events<http://sangoma.com/about_us/events?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>  | Contact<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>  | Wiki<http://wiki.sangoma.com/?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>  | Facebook<http://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>  | Twitter<http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>`|
> | YouTube<http://www.youtube.com/sangomatechnologies?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>
>                 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>
>
>
>
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From juraj.fabo at gmail.com  Sun Oct 23 22:22:22 2011
From: juraj.fabo at gmail.com (Juraj Fabo)
Date: Sun, 23 Oct 2011 20:22:22 +0200
Subject: [Freeswitch-dev] Wanpipe timing source - clock sharing
Message-ID: <CAAqVCyQoAhSFE8+yKzP6L1b4xOK2+R5n2G_NL1LBr1La6vgoJg@mail.gmail.com>

Hi

I'm not sure whether this is a proper forum, but I have an issue with
wanpipe/dahdi driver and the sangoma a102de card.
My problem is probably the timing source change in case when the E1 cable is
disconnected from the card span or spans.
My application plays a PRI_CPE side and uses a testpattern/keepalive sending
on one dedicated e1 channel to detect E1 breakdown.
When there is no cable connected to particular span, keepalive algorithm
send patterns and reads 'dummy' data so it finds out that E1 is broken.
Actually I've read the info from wanpipe.spec about the free run mode in
firmware version V36, I use V37, here is my configuration

sles 11 SP1, wanpipe-3.5.23 (tested also 3.5.18), dahdi-2.5.0.1 (tested also
2.3.0.1), sangoma a102de, HWEC and DTMF detection off.

I have an experience with digium E1 cards and DAHDI driver that removing a
cable from E1 span
makes the span to receive a clock from another span connected to telko.
Or, given there is no other span connected to telko, to use internal clock
on all spans.
So on the digium with dahdi the clock source is moving from span to span as
specified
in "timing source" parameter of /etc/dahdi/system.conf (and indicated as
MASTER in the 1st line of /proc/dahdi/X)

I tested sangoma card together with wanpipe and dahdi drivers. Here the
behaviour was rather different.
When cable was removed, following messages were logged (amongst others)

Oct 21 09:34:06 v184 kernel: [ 3185.691989] wanpipe2: Lost of Signal is
detected!
Oct 21 09:34:10 v184 kernel: [ 3189.604230] wanpipe2:    LOF : ON
Oct 21 09:34:10 v184 kernel: [ 3189.604234] wanpipe2: E1 disconnected!
Oct 21 09:34:10 v184 kernel: [ 3189.604263] wanpipe2: AFT communications
disabled! (Dev Cnt: 2 Cause: Link Down)

Between time 09:34:06 and 09:34:10 my application still writes/reads to the
channel on span which was removed.
After 09:34:10 it is not possible to read/write anymore until the cable is
connected back
 (I use ACE framework event handler registered in reactor with READ | WRITE
| EXCEPT mask)

When last cable is removed (tested also with quad card) then the first span
'recovers' and read/write is possible on it.
Connecting one span 'enables' reading/writing on re-connected span.

I think this might be a timing source issue.
My question is, how to configure wanpipe/dahdi in order to share the clock
on all spans of sangoma,
also when there are spans which are configured and used by an application
but not connected.

Thank you in advance

Juraj Fabo
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From blaisounou at gmail.com  Sun Oct 23 19:13:29 2011
From: blaisounou at gmail.com (marco blaisou)
Date: Sun, 23 Oct 2011 17:13:29 +0200
Subject: [Freeswitch-dev] FreeSWITCH-dev Digest, Vol 64, Issue 19
In-Reply-To: <mailman.7718.1319352469.29749.freeswitch-dev@lists.freeswitch.org>
References: <mailman.7718.1319352469.29749.freeswitch-dev@lists.freeswitch.org>
Message-ID: <CADLgD89euTxBo-qeVLqBhE-x76NBz8gDBGBpwrG6EFGgVTGAkQ@mail.gmail.com>

COMPILING FREESWITCH ON WINDOWS 7 FROM THE GIT
-----------------------------------------------------------------------------------------

Hi guys, just to give you some update from the trenches and give you some
update on
compiling FreeSwitch on Windows from the git.(what a journey!!!)

I can tell you that so far, it appeared to be one of the most frustrating
experience in
building a system from a git: files are missing or are not loaded properly
as they should...

So after a few nights of sctatching my head, here is the status:


*Building on Windows from a git repository
-------------------------------------------------------------*

After git cloning the repository, open vs2008 solution then:

Stop Comodo or any firewall and virus detection
(if not this screws up download scripts giving: util.vbs(184, 2) msxml3.dll
errors)

*0) launch build*

*1) insure all the supposed dowloaded directories are present.
(!!!!!!!!!!!!! VITAL !!!!!!!!)*
Actually there are some problems:

-> issue with opensssl download
-------------------------------------------------
for a strange reason after autodownload  the untar is not complete,
resulting in missing subdirectory
So make sure to do it by hand if necessary

-> issue with CELT download
-------------------------------------------
MUST change prebuilt instructions to :
if not exist "$(ProjectDir)..\celt-0.10.0" cscript /nologo
"$(ProjectDir)util.vbs" GetUnzip
http://files.freeswitch.org/downloads/libs/celt-0.10.0.tar.gz"$(ProjectDir).."
xcopy "$(ProjectDir)\celt\config.h" "$(ProjectDir)..\celt-0.10.0\libcelt\"
/C /D /Y /S /I
xcopy "$(ProjectDir)\celt\float_cast.h"
"$(ProjectDir)..\celt-0.10.0\libcelt\" /C /D /Y /S /I


*2) if you uses Windows sdk 7.1*

edit fake-rfc2533.h and comment out :

#ifdef nodef // *****add this
#ifndef HAVE_STRUCT_ADDRINFO

struct addrinfo {
int    ai_flags;    /* AI_PASSIVE, AI_CANONNAME */
int    ai_family;    /* PF_xxx */
int    ai_socktype;    /* SOCK_xxx */
int    ai_protocol;    /* 0 or IPPROTO_xxx for IPv4 and IPv6 */
size_t    ai_addrlen;    /* length of ai_addr */
char    *ai_canonname;    /* canonical name for hostname */
struct sockaddr *ai_addr;    /* binary address */
struct addrinfo *ai_next;    /* next structure in linked list */
};
#endif /* !HAVE_STRUCT_ADDRINFO */

#endif      // *****add this


*3) if issue with folder: sofia-sip *
cannot include file sip-extra.h ???????????????????????????????????????

(I'm fed up with this so I just decided to get it from a tar)

*4) pb on compile : libsndfile*

in sofia_presence.c
    if (mod_sofia_globals.debug_presence > 1) {
        int i;                    // to be added
        for (i = 0; i < argc; i++) {

//----------------------------------------------------------------------------------

That's it for the time being, most of the stuff (but not ALL) now compile
and I'll try to focus on other pending issues.

OBVIOUSLY ANY ADVICE IS MORE THEN WELCOME !!!!

2011/10/23 <freeswitch-dev-request at lists.freeswitch.org>

> Send FreeSWITCH-dev mailing list submissions to
>        freeswitch-dev at lists.freeswitch.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
>        http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> or, via email, send a message with subject or body 'help' to
>        freeswitch-dev-request at lists.freeswitch.org
>
> You can reach the person managing the list at
>        freeswitch-dev-owner at lists.freeswitch.org
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of FreeSWITCH-dev digest..."
>
> Today's Topics:
>
>   1. == ISSUE building FreeSwitch on Windows (marco blaisou)
>   2. Re: Strange behaviour in SOFIA SIP (Visciano Giovanni)
>   3. Re: == ISSUE building FreeSwitch on Windows (Peter Olsson)
>
>
> ---------- Message transf?r? ----------
> From: marco blaisou <blaisounou at gmail.com>
> To: freeswitch-dev at lists.freeswitch.org
> Date: Sat, 22 Oct 2011 16:59:56 +0200
> Subject: [Freeswitch-dev] == ISSUE building FreeSwitch on Windows
> Hi guys,
> For the few days I've been unsuccesfull in building FreeSwith on Windows 7
>
> I did follow the Wiki line per line, but some thing is really going wrong:
> > apparently the files in /libs directory are not loaded from the net (ie
> the util.vbs does seem to load and unpack the libs)
> error is : *freeswitch\libs\win32\util.vbs(184, 2) msxml3.dll: Le syst?me
> ne trouve pas la ressource sp?cifi?e (ie system doesn't find specified
> resource)*
>
> > then obviously the compiler starts complaining about missing stuff such
> as:
> *c:\users\jc\desktop\voip\freeswitch\freeswitch\src\mod\endpoints\mod_sofia\mod_sofia.h(47)
> : fatal error C1083: Cannot open include file: 'switch_version.h': No such
> file or directory*
>
> To summarize:
> I loaded the FreeSwitch git repository with TortoiseGit
>
> My config is :
> Windows 7/64bits
> VSC2008 sp1
> Windows SDK 7.1
>
> My firewall is desactivated as suggested
>
> I did follow the recommendation to clear some files:
> *del libs\apr\include\apr.h
> del libs\js\config.h
> del libs\js\src\jsautocfg.h
> del libs\js\nsprpub\pr\include\prcpucfg.h
> del libs\iksemel\include\config.h
> del libs\xmlrpc-c\xmlrpc_config.h
> del libs\libsndfile\src\sfconfig.h
> del libs\sofia-sip\libsofia-sip-ua\su\sofia-sip\su_configure.h
> del libs\js\nsprpub\pr\include\prcpucfg.h
>
> cd libs\win32\libsndfile
> rename config.h sfconfig.h*
>
> Could a guru provide some hints on how to solve this???
> Thanks
>
>
>
>
> ---------- Message transf?r? ----------
> From: "Visciano Giovanni" <Giovanni.Visciano at italtel.it>
> To: <freeswitch-dev at lists.freeswitch.org>
> Date: Sat, 22 Oct 2011 20:30:06 +0200
> Subject: Re: [Freeswitch-dev] Strange behaviour in SOFIA SIP
> Now I understand.
> If both legs use the same codec, the codec ("fake") will be used in a
> pass-through way,
> so no additional transcoding is applied, as in PROXY PASS-THROUGH
> condition.
>
> So let me reason in a default media scenario (no proxy-media).
>
> Now, as I explained in the first mail, sofia build "ep_codec_string"
> (SWITCH_
> ORIGINATOR_CODEC_VARIABLE) as the intersection between the codec list
> received
> in SDP offer and the codec loaded in the core.
> So if the offered SDP codec list is NOT a subset of the loaded core codecs
> the other
> legs will miss some codecs proposed by the calling endpoint.
> That said, if I want to be sure every codecs my endpoint support won't be
> filtered I
> have to register a "fake" pass-through codec for each one.
>
> ... you did not answer to my last question :).
> I haven't found any other endpoint module the can interwork in PROXY mode
> with
> sofia (even h323 doesn't support this). My question: is there a standard/
> recommended way to exchange capability between endpoint of different nature
> in FS
> proxy mode? Maybe using SDP o/a as done in sofia sip?
>
> Thank you again for you help.
> Giovanni
>
>
> -----Original Message-----
> From: freeswitch-dev-bounces at lists.freeswitch.org on behalf of Anthony
> Minessale
> Sent: Sat 10/22/2011 4:48 PM
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] Strange behaviour in SOFIA SIP
>
> that's what I was trying to explain.
> PROXY mode does not let you manipulate anything, it's pure
> pass-through negotiation between the endpoints.
>
> All the stuff with ep_codec_string are relevant only when you are not
> trying to use proxy mode.
> This is a common mistake people make assuming not using proxy mode
> means somehow doing more work but it's not the case, any time both
> legs of a call use the same codec it's equally efficient to what you
> end up with in proxy mode.  proxy mode simply means take the inbound
> SDP and offer it AS IS to the far end and vice-versa and use a fake
> codec in the middle that is agnostic.
>
>
>
> On Fri, Oct 21, 2011 at 10:36 AM, Visciano Giovanni
> <Giovanni.Visciano at italtel.it> wrote:
> > Hello Anthony.
> > Maybe I want not so clear, let me explain.
> >
> > I do not want to apply any codec manipulation, I'm working on a prototype
> endpoint module that has "proxy media capabilty" and wants to interoperate
> with other endpoints.
> > What I saw is that in a sofia vs sofia call (proxy mode+late negotiation)
> they exchange through FS calling/called endpoint capability looking at SDP
> session carried by channel variables, so internally they use SDP O/A scheme.
> > I thought that "ep_codec_string"->SWITCH_ORIGINATOR_CODEC_VARIABLE could
> be used to exchange capability without using a protocol specific format
> (SDP).
> > If this is not the case, what's the right way to negotiate capability
> between endpoints in proxy mode?
> >
> > Than you.
> > Giovanni
> >
> > PS: I did not find any existing endpoint except sofia that support this.
> Am I right?
> >
> > -----Original Message-----
> > From: freeswitch-dev-bounces at lists.freeswitch.org on behalf of Anthony
> Minessale
> > Sent: Fri 10/21/2011 5:06 PM
> > To: freeswitch-dev at lists.freeswitch.org
> > Subject: Re: [Freeswitch-dev] Strange behaviour in SOFIA SIP
> >
> > any codec manipulation techniques and proxy media mode are not
> > compatible.  Proxy media mode means stay completely out of the codec
> > negotiation process.
> >
> > On Thu, Oct 20, 2011 at 3:11 PM, Visciano Giovanni
> > <Giovanni.Visciano at italtel.it> wrote:
> >> Preconditions on sip profile
> >> ----------------------------
> >> inbound-codec-prefs NOT set
> >> outbound-codec-prefs NOT set
> >> inbound-proxy-media=TRUE
> >> inbound-bypass-media=FALSE
> >> inbound-late-negotiation=TRUE + inherit_codec=TRUE
> >>
> >> TEST scenario
> >> -------------
> >> SIP phone send INVITE with SDP to FS
> >>
> >> INVITE
> >> SDP[codelist=PCMA, PCMU, G722, G729] --> FS
> >>
> >> FS has just the CORE_PC_MODULE [PCMA, PCMU, PASS-THR, VIDEO-PASS-THR,
> L16]
> >>
> >> The "ep_codec_string" channel variable is loaded with [PCMA, PCMU]
> >>
> >> EXPECTED
> >> --------
> >> In "proxy-media + late-negotiation" scenario:
> >> ep_codec_string=[PCMA,PCMU,G722,G729]
> >>
> >> PROBLEM
> >> -------
> >> According to the documentation
> >> (http://wiki.freeswitch.org/wiki/Codec_negotiation):
> >>
> >> "ep_codec_string: This variable is only available if late negotiation is
> >> enabled on the profile. It's a readable string containing all the codecs
> >> proposed by the calling endpoint."
> >> and
> >> "inherit_codec=true (only applicable when late-negotiation is enabled)
> will
> >> take the codec negotiated when the B leg answers and pass it to the A
> leg so
> >> it also uses the same codec (if the A leg supports it); otherwise, it
> will
> >> use whatever it can from its own list"
> >>
> >> The variable ep_codec_string doesn't contain ALL the codecs proposed by
> the
> >> calling endpoint. It contains the intersection between the codecs
> proposed
> >> by the calling and the codecs loaded in FS.
> >> In proxy-media / no-media configuration the called endpoint doesn't have
> the
> >> calling codec list so he can't negotiate using all the capabilities
> >> available.
> >> (I'm working on a test endpoint module that should work in proxy and
> >> no-media configuration ... mod_sofia vs mod_sofia works because they
> look at
> >> the remote sdp variable)
> >>
> >> Source reference - sofia_glue.c->sofia_glue_set_r_sdp_codec_string()
> >>
> >>
> >> Regards,
> >> Giovanni
> >>
> >>
> >>
> >> Internet Email Confidentiality Footer
> >>
> >>
> ********************************************************************************************************************************************
> >>
> >> La presente comunicazione, con le informazioni in essa contenute e ogni
> >> documento o file allegato, e' rivolta unicamente alla/e persona/e cui e'
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> C.P.,
> >> D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali).
> Se
> >> avete ricevuto questa comunicazione per errore, vi preghiamo di darne
> >> immediata notizia al mittente e di distruggere il messaggio originale e
> ogni
> >> file allegato senza farne copia alcuna o riprodurne in alcun modo il
> >> contenuto.
> >>
> >> This e-mail and its attachments are intended for the addressee(s) only
> and
> >> are confidential and/or may contain legally privileged information. If
> you
> >> have received this message by mistake or are not one of the addressees
> >> above, you may take no action based on it, and you may not copy or show
> it
> >> to anyone; please reply to this e-mail and point out the error which has
> >> occurred.
> >>
> ********************************************************************************************************************************************
> >>
> >> _______________________________________________
> >> Join us at ClueCon 2011, Aug 9-11, Chicago
> >> http://www.cluecon.com 877-7-4ACLUE
> >>
> >> 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
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > 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
> >
> >
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > 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
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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
>
>
>
>
> ---------- Message transf?r? ----------
> From: Peter Olsson <peter.olsson at visionutveckling.se>
> To: "freeswitch-dev at lists.freeswitch.org" <
> freeswitch-dev at lists.freeswitch.org>
> Date: Sun, 23 Oct 2011 08:42:42 +0200
> Subject: Re: [Freeswitch-dev] == ISSUE building FreeSwitch on Windows
> I believe that you might be missing some system dll's on your machine.
>
> msxml3.dll is something that should exist on your computer, and I think the
> download script is depending on it.
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [
> freeswitch-dev-bounces at lists.freeswitch.org] f&#246;r marco blaisou [
> blaisounou at gmail.com]
> Skickat: den 22 oktober 2011 16:59
> Till: freeswitch-dev at lists.freeswitch.org
> ?mne: [Freeswitch-dev] == ISSUE building FreeSwitch on Windows
>
> Hi guys,
> For the few days I've been unsuccesfull in building FreeSwith on Windows 7
>
> I did follow the Wiki line per line, but some thing is really going wrong:
> > apparently the files in /libs directory are not loaded from the net (ie
> the util.vbs does seem to load and unpack the libs)
> error is : freeswitch\libs\win32\util.vbs(184, 2) msxml3.dll: Le syst?me ne
> trouve pas la ressource sp?cifi?e (ie system doesn't find specified
> resource)
>
> > then obviously the compiler starts complaining about missing stuff such
> as:
> c:\users\jc\desktop\voip\freeswitch\freeswitch\src\mod\endpoints\mod_sofia\mod_sofia.h(47)
> : fatal error C1083: Cannot open include file: 'switch_version.h': No such
> file or directory
>
> To summarize:
> I loaded the FreeSwitch git repository with TortoiseGit
>
> My config is :
> Windows 7/64bits
> VSC2008 sp1
> Windows SDK 7.1
>
> My firewall is desactivated as suggested
>
> I did follow the recommendation to clear some files:
> del libs\apr\include\apr.h
> del libs\js\config.h
> del libs\js\src\jsautocfg.h
> del libs\js\nsprpub\pr\include\prcpucfg.h
> del libs\iksemel\include\config.h
> del libs\xmlrpc-c\xmlrpc_config.h
> del libs\libsndfile\src\sfconfig.h
> del libs\sofia-sip\libsofia-sip-ua\su\sofia-sip\su_configure.h
> del libs\js\nsprpub\pr\include\prcpucfg.h
>
> cd libs\win32\libsndfile
> rename config.h sfconfig.h
>
> Could a guru provide some hints on how to solve this???
> Thanks
>
>
> !DSPAM:4ea3271432761137115228!
>
>
>
> _______________________________________________
> 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 blaisounou at gmail.com  Sun Oct 23 19:52:39 2011
From: blaisounou at gmail.com (marco blaisou)
Date: Sun, 23 Oct 2011 17:52:39 +0200
Subject: [Freeswitch-dev] unresolved external symbol _TPTAG_CAPT and
	_NTATAG_SRV_503 ?
Message-ID: <CADLgD89yJuTZLK3TjxSX405M-aRZe9KBA+6vXZk_SK=fYLDyYw@mail.gmail.com>

as a follow up of previous mail

21>   Creating library Win32\Release/mod_sofia.2008.lib and object
Win32\Release/mod_sofia.2008.exp
21>mod_sofia.obj : error LNK2001: unresolved external symbol _TPTAG_CAPT
21>sofia.obj : error LNK2001: unresolved external symbol _NTATAG_SRV_503
21>C:\Users\jc\Desktop\VOIP\freeswitch\freeswitch\Release/mod/mod_sofia.dll
: fatal error LNK1120: 2 unresolved externals
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From moises.silva at gmail.com  Mon Oct 24 08:04:17 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Mon, 24 Oct 2011 00:04:17 -0400
Subject: [Freeswitch-dev] FreeTDM delay reconnecting E1/T1
In-Reply-To: <CAPLGU-4B2oyU+i6B0d09=GULe0kUViYaYW1EVPSotR45oAr0ig@mail.gmail.com>
References: <CAPLGU-71UcZxBD=M1CccVO0xYwNsRqH5Kk0KETyxZdmmHFtDvQ@mail.gmail.com>
	<CAA4nhyCs5rN54kUAz27RVCRY=fGGkshTHE96wtSt8oDFuF0bCg@mail.gmail.com>
	<CAPLGU-4xAjjgGWXtZHa+ukJ9vLZ0QtNz-knwmfoPc6gmngV6Aw@mail.gmail.com>
	<CAA4nhyCOBei_aEmtDHqeFAYiSztNpYmncv993ZgwF744GGHYEg@mail.gmail.com>
	<CAPLGU-6SPvDB3=SyPddpDpjZ-1OaeAcX7YdhQPdJYxgyRKHNog@mail.gmail.com>
	<CAPLGU-4B2oyU+i6B0d09=GULe0kUViYaYW1EVPSotR45oAr0ig@mail.gmail.com>
Message-ID: <CAA4nhyBmqy2SKUbZRVtA9Go8677CtD8VoKP=UpqFPDsF2X4PNg@mail.gmail.com>

On Thu, Oct 20, 2011 at 10:21 AM, Glaucio Vaz de Mello <gvmello at yx.cl>wrote:

> Hello,
>
> Just to tell you that I tested it again with latest Git source and latest
> sangoma driver.
> Delay still happens, but are less than before.
>
> When E1/T1 cable is disconnected, wanpipemon command shows "Loss of Signal"
> in less than 1 second.
> Meanwhile, fs_cli "ftdm dump" still shows "physical status" as "ok" for 4
> or 5 seconds, then it turns to "alarmed".
>
> Reconnecting the cable I experienced bigger delays: while wanpipemon shows
> "Loss of Signal OFF" in less than 1 second, fs_cli "ftdm dump" takes more
> than 20 seconds to became "ok" again.
>
> If you think this should be fixed and need more information, logs or ssh
> access, just let me know and I'll be glad to provide it.
>

The "loss of signal" indicator is a very low level indication that there is
sudden disconnection, but the T1/E1 standard has a timer to indicate red
alarm, 4 seconds sounds about right top off my head. However 20 seconds to
recover from red alarm is too much. Please contact techdesk at sangoma.com to
have a support engineer take a closer look at your problem.

Feel free to ping me on IRC as well.


*Moises Silva
**Software Engineer, Development Manager***

msilva 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 x128

f.   +1 905 474 9223



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From peter.olsson at visionutveckling.se  Mon Oct 24 08:56:17 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 24 Oct 2011 06:56:17 +0200
Subject: [Freeswitch-dev] FreeSWITCH-dev Digest, Vol 64, Issue 19
In-Reply-To: <CADLgD89euTxBo-qeVLqBhE-x76NBz8gDBGBpwrG6EFGgVTGAkQ@mail.gmail.com>
References: <mailman.7718.1319352469.29749.freeswitch-dev@lists.freeswitch.org>,
	<CADLgD89euTxBo-qeVLqBhE-x76NBz8gDBGBpwrG6EFGgVTGAkQ@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF177@cooper>

I build FS on Windows every day, I rarely have problems with it, and if a problem occurs it's mostly because of a change in the code that works on Linux, but not Windows - but these kind of problems are usually fixed later the same day (or i just submit a patch, and it will be applied soon enough).

Last time I did a complete rebuild of FS (with no libs downloaded) I also found some problems with openssl (but this has only happened once), but I just removed the openssl-folder and rebuilt, and everything worked, the rest of the things you mention here I've never heard of before. My guess is you have problems related to antivirus or UAC on your machine, but I will try to do a complete new rebuild today, and see how it looks for me - I'll get back to you after that.

Also, please don't hijack other threads, send a new mail the next time, instead of replying to a digest mail.

/Peter
________________________________________
Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [freeswitch-dev-bounces at lists.freeswitch.org] f&#246;r marco blaisou [blaisounou at gmail.com]
Skickat: den 23 oktober 2011 17:13
Till: freeswitch-dev at lists.freeswitch.org
?mne: Re: [Freeswitch-dev] FreeSWITCH-dev Digest, Vol 64, Issue 19

COMPILING FREESWITCH ON WINDOWS 7 FROM THE GIT
-----------------------------------------------------------------------------------------

Hi guys, just to give you some update from the trenches and give you some update on
compiling FreeSwitch on Windows from the git.(what a journey!!!)

I can tell you that so far, it appeared to be one of the most frustrating experience in
building a system from a git: files are missing or are not loaded properly as they should...

So after a few nights of sctatching my head, here is the status:


Building on Windows from a git repository
-------------------------------------------------------------

After git cloning the repository, open vs2008 solution then:

Stop Comodo or any firewall and virus detection
(if not this screws up download scripts giving: util.vbs(184, 2) msxml3.dll errors)

0) launch build

1) insure all the supposed dowloaded directories are present. (!!!!!!!!!!!!! VITAL !!!!!!!!)
Actually there are some problems:

-> issue with opensssl download
-------------------------------------------------
for a strange reason after autodownload  the untar is not complete, resulting in missing subdirectory
So make sure to do it by hand if necessary

-> issue with CELT download
-------------------------------------------
MUST change prebuilt instructions to :
if not exist "$(ProjectDir)..\celt-0.10.0" cscript /nologo "$(ProjectDir)util.vbs" GetUnzip http://files.freeswitch.org/downloads/libs/celt-0.10.0.tar.gz "$(ProjectDir).."
xcopy "$(ProjectDir)\celt\config.h" "$(ProjectDir)..\celt-0.10.0\libcelt\" /C /D /Y /S /I
xcopy "$(ProjectDir)\celt\float_cast.h" "$(ProjectDir)..\celt-0.10.0\libcelt\" /C /D /Y /S /I


2) if you uses Windows sdk 7.1

edit fake-rfc2533.h and comment out :

#ifdef nodef // *****add this
#ifndef HAVE_STRUCT_ADDRINFO

struct addrinfo {
int    ai_flags;    /* AI_PASSIVE, AI_CANONNAME */
int    ai_family;    /* PF_xxx */
int    ai_socktype;    /* SOCK_xxx */
int    ai_protocol;    /* 0 or IPPROTO_xxx for IPv4 and IPv6 */
size_t    ai_addrlen;    /* length of ai_addr */
char    *ai_canonname;    /* canonical name for hostname */
struct sockaddr *ai_addr;    /* binary address */
struct addrinfo *ai_next;    /* next structure in linked list */
};
#endif /* !HAVE_STRUCT_ADDRINFO */

#endif      // *****add this


3) if issue with folder: sofia-sip
cannot include file sip-extra.h ???????????????????????????????????????

(I'm fed up with this so I just decided to get it from a tar)

4) pb on compile : libsndfile

in sofia_presence.c
    if (mod_sofia_globals.debug_presence > 1) {
        int i;                    // to be added
        for (i = 0; i < argc; i++) {

//----------------------------------------------------------------------------------

That's it for the time being, most of the stuff (but not ALL) now compile
and I'll try to focus on other pending issues.

OBVIOUSLY ANY ADVICE IS MORE THEN WELCOME !!!!

2011/10/23 <freeswitch-dev-request at lists.freeswitch.org<mailto:freeswitch-dev-request at lists.freeswitch.org>>
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When replying, please edit your Subject line so it is more specific
than "Re: Contents of FreeSWITCH-dev digest..."

Today's Topics:

  1. == ISSUE building FreeSwitch on Windows (marco blaisou)
  2. Re: Strange behaviour in SOFIA SIP (Visciano Giovanni)
  3. Re: == ISSUE building FreeSwitch on Windows (Peter Olsson)


---------- Message transf?r? ----------
From: marco blaisou <blaisounou at gmail.com<mailto:blaisounou at gmail.com>>
To: freeswitch-dev at lists.freeswitch.org<mailto:freeswitch-dev at lists.freeswitch.org>
Date: Sat, 22 Oct 2011 16:59:56 +0200
Subject: [Freeswitch-dev] == ISSUE building FreeSwitch on Windows
Hi guys,
For the few days I've been unsuccesfull in building FreeSwith on Windows 7

I did follow the Wiki line per line, but some thing is really going wrong:
> apparently the files in /libs directory are not loaded from the net (ie the util.vbs does seem to load and unpack the libs)
error is : freeswitch\libs\win32\util.vbs(184, 2) msxml3.dll: Le syst?me ne trouve pas la ressource sp?cifi?e (ie system doesn't find specified resource)

> then obviously the compiler starts complaining about missing stuff such as:
c:\users\jc\desktop\voip\freeswitch\freeswitch\src\mod\endpoints\mod_sofia\mod_sofia.h(47) : fatal error C1083: Cannot open include file: 'switch_version.h': No such file or directory

To summarize:
I loaded the FreeSwitch git repository with TortoiseGit

My config is :
Windows 7/64bits
VSC2008 sp1
Windows SDK 7.1

My firewall is desactivated as suggested

I did follow the recommendation to clear some files:
del libs\apr\include\apr.h
del libs\js\config.h
del libs\js\src\jsautocfg.h
del libs\js\nsprpub\pr\include\prcpucfg.h
del libs\iksemel\include\config.h
del libs\xmlrpc-c\xmlrpc_config.h
del libs\libsndfile\src\sfconfig.h
del libs\sofia-sip\libsofia-sip-ua\su\sofia-sip\su_configure.h
del libs\js\nsprpub\pr\include\prcpucfg.h

cd libs\win32\libsndfile
rename config.h sfconfig.h

Could a guru provide some hints on how to solve this???
Thanks




---------- Message transf?r? ----------
From: "Visciano Giovanni" <Giovanni.Visciano at italtel.it<mailto:Giovanni.Visciano at italtel.it>>
To: <freeswitch-dev at lists.freeswitch.org<mailto:freeswitch-dev at lists.freeswitch.org>>
Date: Sat, 22 Oct 2011 20:30:06 +0200
Subject: Re: [Freeswitch-dev] Strange behaviour in SOFIA SIP
Now I understand.
If both legs use the same codec, the codec ("fake") will be used in a pass-through way,
so no additional transcoding is applied, as in PROXY PASS-THROUGH condition.

So let me reason in a default media scenario (no proxy-media).

Now, as I explained in the first mail, sofia build "ep_codec_string" (SWITCH_
ORIGINATOR_CODEC_VARIABLE) as the intersection between the codec list received
in SDP offer and the codec loaded in the core.
So if the offered SDP codec list is NOT a subset of the loaded core codecs the other
legs will miss some codecs proposed by the calling endpoint.
That said, if I want to be sure every codecs my endpoint support won't be filtered I
have to register a "fake" pass-through codec for each one.

... you did not answer to my last question :).
I haven't found any other endpoint module the can interwork in PROXY mode with
sofia (even h323 doesn't support this). My question: is there a standard/
recommended way to exchange capability between endpoint of different nature in FS
proxy mode? Maybe using SDP o/a as done in sofia sip?

Thank you again for you help.
Giovanni


-----Original Message-----
From: freeswitch-dev-bounces at lists.freeswitch.org<mailto:freeswitch-dev-bounces at lists.freeswitch.org> on behalf of Anthony Minessale
Sent: Sat 10/22/2011 4:48 PM
To: freeswitch-dev at lists.freeswitch.org<mailto:freeswitch-dev at lists.freeswitch.org>
Subject: Re: [Freeswitch-dev] Strange behaviour in SOFIA SIP

that's what I was trying to explain.
PROXY mode does not let you manipulate anything, it's pure
pass-through negotiation between the endpoints.

All the stuff with ep_codec_string are relevant only when you are not
trying to use proxy mode.
This is a common mistake people make assuming not using proxy mode
means somehow doing more work but it's not the case, any time both
legs of a call use the same codec it's equally efficient to what you
end up with in proxy mode.  proxy mode simply means take the inbound
SDP and offer it AS IS to the far end and vice-versa and use a fake
codec in the middle that is agnostic.



On Fri, Oct 21, 2011 at 10:36 AM, Visciano Giovanni
<Giovanni.Visciano at italtel.it<mailto:Giovanni.Visciano at italtel.it>> wrote:
> Hello Anthony.
> Maybe I want not so clear, let me explain.
>
> I do not want to apply any codec manipulation, I'm working on a prototype endpoint module that has "proxy media capabilty" and wants to interoperate with other endpoints.
> What I saw is that in a sofia vs sofia call (proxy mode+late negotiation) they exchange through FS calling/called endpoint capability looking at SDP session carried by channel variables, so internally they use SDP O/A scheme.
> I thought that "ep_codec_string"->SWITCH_ORIGINATOR_CODEC_VARIABLE could be used to exchange capability without using a protocol specific format (SDP).
> If this is not the case, what's the right way to negotiate capability between endpoints in proxy mode?
>
> Than you.
> Giovanni
>
> PS: I did not find any existing endpoint except sofia that support this. Am I right?
>
> -----Original Message-----
> From: freeswitch-dev-bounces at lists.freeswitch.org<mailto:freeswitch-dev-bounces at lists.freeswitch.org> on behalf of Anthony Minessale
> Sent: Fri 10/21/2011 5:06 PM
> To: freeswitch-dev at lists.freeswitch.org<mailto:freeswitch-dev at lists.freeswitch.org>
> Subject: Re: [Freeswitch-dev] Strange behaviour in SOFIA SIP
>
> any codec manipulation techniques and proxy media mode are not
> compatible.  Proxy media mode means stay completely out of the codec
> negotiation process.
>
> On Thu, Oct 20, 2011 at 3:11 PM, Visciano Giovanni
> <Giovanni.Visciano at italtel.it<mailto:Giovanni.Visciano at italtel.it>> wrote:
>> Preconditions on sip profile
>> ----------------------------
>> inbound-codec-prefs NOT set
>> outbound-codec-prefs NOT set
>> inbound-proxy-media=TRUE
>> inbound-bypass-media=FALSE
>> inbound-late-negotiation=TRUE + inherit_codec=TRUE
>>
>> TEST scenario
>> -------------
>> SIP phone send INVITE with SDP to FS
>>
>> INVITE
>> SDP[codelist=PCMA, PCMU, G722, G729] --> FS
>>
>> FS has just the CORE_PC_MODULE [PCMA, PCMU, PASS-THR, VIDEO-PASS-THR, L16]
>>
>> The "ep_codec_string" channel variable is loaded with [PCMA, PCMU]
>>
>> EXPECTED
>> --------
>> In "proxy-media + late-negotiation" scenario:
>> ep_codec_string=[PCMA,PCMU,G722,G729]
>>
>> PROBLEM
>> -------
>> According to the documentation
>> (http://wiki.freeswitch.org/wiki/Codec_negotiation):
>>
>> "ep_codec_string: This variable is only available if late negotiation is
>> enabled on the profile. It's a readable string containing all the codecs
>> proposed by the calling endpoint."
>> and
>> "inherit_codec=true (only applicable when late-negotiation is enabled) will
>> take the codec negotiated when the B leg answers and pass it to the A leg so
>> it also uses the same codec (if the A leg supports it); otherwise, it will
>> use whatever it can from its own list"
>>
>> The variable ep_codec_string doesn't contain ALL the codecs proposed by the
>> calling endpoint. It contains the intersection between the codecs proposed
>> by the calling and the codecs loaded in FS.
>> In proxy-media / no-media configuration the called endpoint doesn't have the
>> calling codec list so he can't negotiate using all the capabilities
>> available.
>> (I'm working on a test endpoint module that should work in proxy and
>> no-media configuration ... mod_sofia vs mod_sofia works because they look at
>> the remote sdp variable)
>>
>> Source reference - sofia_glue.c->sofia_glue_set_r_sdp_codec_string()
>>
>>
>> Regards,
>> Giovanni
>>
>>
>>
>> Internet Email Confidentiality Footer
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>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
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>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
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> 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>
>
> _______________________________________________
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>
>
> _______________________________________________
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>
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>



--
Anthony Minessale II

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

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

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---------- Message transf?r? ----------
From: Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>>
To: "freeswitch-dev at lists.freeswitch.org<mailto:freeswitch-dev at lists.freeswitch.org>" <freeswitch-dev at lists.freeswitch.org<mailto:freeswitch-dev at lists.freeswitch.org>>
Date: Sun, 23 Oct 2011 08:42:42 +0200
Subject: Re: [Freeswitch-dev] == ISSUE building FreeSwitch on Windows
I believe that you might be missing some system dll's on your machine.

msxml3.dll is something that should exist on your computer, and I think the download script is depending on it.

/Peter
________________________________________
Fr?n: freeswitch-dev-bounces at lists.freeswitch.org<mailto:freeswitch-dev-bounces at lists.freeswitch.org> [freeswitch-dev-bounces at lists.freeswitch.org<mailto:freeswitch-dev-bounces at lists.freeswitch.org>] f&#246;r marco blaisou [blaisounou at gmail.com<mailto:blaisounou at gmail.com>]
Skickat: den 22 oktober 2011 16:59
Till: freeswitch-dev at lists.freeswitch.org<mailto:freeswitch-dev at lists.freeswitch.org>
?mne: [Freeswitch-dev] == ISSUE building FreeSwitch on Windows

Hi guys,
For the few days I've been unsuccesfull in building FreeSwith on Windows 7

I did follow the Wiki line per line, but some thing is really going wrong:
> apparently the files in /libs directory are not loaded from the net (ie the util.vbs does seem to load and unpack the libs)
error is : freeswitch\libs\win32\util.vbs(184, 2) msxml3.dll: Le syst?me ne trouve pas la ressource sp?cifi?e (ie system doesn't find specified resource)

> then obviously the compiler starts complaining about missing stuff such as:
c:\users\jc\desktop\voip\freeswitch\freeswitch\src\mod\endpoints\mod_sofia\mod_sofia.h(47) : fatal error C1083: Cannot open include file: 'switch_version.h': No such file or directory

To summarize:
I loaded the FreeSwitch git repository with TortoiseGit

My config is :
Windows 7/64bits
VSC2008 sp1
Windows SDK 7.1

My firewall is desactivated as suggested

I did follow the recommendation to clear some files:
del libs\apr\include\apr.h
del libs\js\config.h
del libs\js\src\jsautocfg.h
del libs\js\nsprpub\pr\include\prcpucfg.h
del libs\iksemel\include\config.h
del libs\xmlrpc-c\xmlrpc_config.h
del libs\libsndfile\src\sfconfig.h
del libs\sofia-sip\libsofia-sip-ua\su\sofia-sip\su_configure.h
del libs\js\nsprpub\pr\include\prcpucfg.h

cd libs\win32\libsndfile
rename config.h sfconfig.h

Could a guru provide some hints on how to solve this???
Thanks






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!DSPAM:4ea488b232761113570065!


From xin at ind.rwth-aachen.de  Mon Oct 24 15:36:26 2011
From: xin at ind.rwth-aachen.de (Han Xin)
Date: Mon, 24 Oct 2011 13:36:26 +0200
Subject: [Freeswitch-dev] trouble on how to remove the media bug fromthe
 session after the sessions are closed
In-Reply-To: <E8A95E8A-E508-4ED1-8CFB-D195DC27C8AB@avgs.ca>
Message-ID: <f339c9da3803750d18192a2082974006@gw.ind.rwth-aachen.de>

 Awesome! Problem solved. Thank you all!
 
----------------urspr?ngliche Nachricht-----------------
Von: "Mathieu Rene" mrene_lists at avgs.ca 
An: freeswitch-dev at lists.freeswitch.org 
Datum: Thu, 20 Oct 2011 23:39:33 -0400
-------------------------------------------------
 
 
> Hi,
> 
> On line 188, you call switch_core_session_get_partner. That function 
> internally calls switch_core_session_locate(), you have to rwunlock it.
> 
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca 
> 
> 
> 
> 
> On 2011-10-20, at 12:09 PM, Han Xin wrote:
> 
>> I tried a lot in the past days but still got no progress. So I upload my code here 
>> to ask for help again.
>> 
>> Thanks anyway!
>> 
>> ----------------original message-----------------
>> From: "Anthony Minessale" anthony.minessale at gmail.com 
>> To: freeswitch-dev at lists.freeswitch.org 
>> Date: Mon, 17 Oct 2011 09:39:26 -0500
>> -------------------------------------------------
>> 
>> 
>>> I have no more guesses. You need to review your custom code and find
>>> out where you read locked your session.
>>> 
>>> 
>>> On Mon, Oct 17, 2011 at 2:38 AM, Han Xin xin at ind.rwth-aachen.de wrote:
>>>> no, no use of ivr_originate function.
>>>> I constructed a helper struct which include the remote session, could 
>>>> this
>>>> cause the session releasing problem?
>>>> 
>>>> ----------------urspr?ngliche Nachricht-----------------
>>>> Von: "Anthony Minessale" anthony.minessale at gmail.com 
>>>> An: freeswitch-dev at lists.freeswitch.org 
>>>> Datum: Thu, 13 Oct 2011 15:55:35 -0500
>>>> -------------------------------------------------
>>>> 
>>>> 
>>>>> how about switch_ivr_originate() ?
>>>>> 
>>>>> On Thu, Oct 13, 2011 at 2:19 AM, Han Xin xin at ind.rwth-aachen.de 
>>>>> wrote:
>>>>>> yes I use a custom mod and the latest git on Oct 10 but I did not use 
>>>>>> the
>>>>>> function
>>>>>> "switch_core_session_locate".
>>>>>> 
>>>>>> ----------------urspr?ngliche 
>>>>>> Nachricht-----------------
>>>>>> Von: "Anthony Minessale" anthony.minessale at gmail.com 
>>>>>> An: freeswitch-dev at lists.freeswitch.org 
>>>>>> Datum: Wed, 12 Oct 2011 17:06:41 -0500
>>>>>> -------------------------------------------------
>>>>>> 
>>>>>> 
>>>>>>> looks like you have some custom mods maybe you used
>>>>>>> switch_core_session_locate and did not rwunlock it?
>>>>>>> are you using latest GIT ?
>>>>>>> 
>>>>>>> 
>>>>>>> On Mon, Oct 10, 2011 at 11:03 AM, Han Xin 
>>>>>>> xin at ind.rwth-aachen.de 
>>>>>>> wrote:
>>>>>>>> yes, I use the SWITCH_ABC_TYPE_CLOSE type and I've 
>>>>>>>> found the
>>>>>>>> media
>>>>>>>> bugs are
>>>>>>>> automatically removed with the DEBUG log infomation. 
>>>>>>>> Now the
>>>>>>>> problem
>>>>>>>> is the
>>>>>>>> session ending.
>>>>>>>> 
>>>>>>>> Detailed situation description: I added two media bugs 
>>>>>>>> to both
>>>>>>>> local
>>>>>>>> session
>>>>>>>> and remote session of one call. When I ended the call, the 
>>>>>>>> media
>>>>>>>> bugs are
>>>>>>>> removed
>>>>>>>> first, and then the session should be ended. But only the 
>>>>>>>> local
>>>>>>>> session
>>>>>>>> is ended
>>>>>>>> correctly.
>>>>>>>> When I press F3 and F4 to show current channels and calls, 
>>>>>>>> the
>>>>>>>> remote
>>>>>>>> channel
>>>>>>>> and call info is till there. But when I try to dump uuid to 
>>>>>>>> get
>>>>>>>> channel
>>>>>>>> infomation
>>>>>>>> with the cmd "uuid_dump remote_session_uuid", it says 
>>>>>>>> there
>>>>>>>> is no
>>>>>>>> such
>>>>>>>> channel!
>>>>>>>> 
>>>>>>>> here is some related log info:
>>>>>>>> 
>>>>>>>> 2011-10-10 17:36:41.428463 [ERR] mod_bwe.c:345
>>>>>>>> 28b3a542-aecd-4b9b-9f55-3c2674a25b5c: Remove 
>>>>>>>> local
>>>>>>>> variable
>>>>>>>> store!
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_ivr_bridge.c:1332
>>>>>>>> sofia/internal/sip:1001 at 137.226.198.252 :5060 
>>>>>>>> skip
>>>>>>>> receive
>>>>>>>> message
>>>>>>>> [UNBRIDGE] (channel is hungup already)
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_ivr_bridge.c:1335
>>>>>>>> sofia/internal/1000 at asterisk skip receive message
>>>>>>>> [UNBRIDGE]
>>>>>>>> (channel is
>>>>>>>> hungup already)
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_media_bug.c:480
>>>>>>>> Removing BUG from
>>>>>>>> sofia/internal/sip:1001 at 137.226.198.252 
>>>>>>>> :5060
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:602
>>>>>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) 
>>>>>>>> State
>>>>>>>> HANGUP
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG] 
>>>>>>>> mod_sofia.c:454
>>>>>>>> sofia/internal/sip:1001 at 137.226.198.252 :5060
>>>>>>>> Overriding SIP
>>>>>>>> cause 480
>>>>>>>> with 200 from the other leg
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG] 
>>>>>>>> mod_sofia.c:460
>>>>>>>> Channel
>>>>>>>> sofia/internal/sip:1001 at 137.226.198.252 :5060 
>>>>>>>> hanging
>>>>>>>> up,
>>>>>>>> cause:
>>>>>>>> NORMAL_CLEARING
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG] 
>>>>>>>> mod_sofia.c:503
>>>>>>>> Sending BYE
>>>>>>>> to
>>>>>>>> sofia/internal/sip:1001 at 137.226.198.252 :5060
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_session.c:2235
>>>>>>>> sofia/internal/1000 at asterisk skip receive message
>>>>>>>> [APPLICATION_EXEC_COMPLETE] (channel is hungup 
>>>>>>>> already)
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:417
>>>>>>>> (sofia/internal/1000 at asterisk ) State EXECUTE going 
>>>>>>>> to
>>>>>>>> sleep
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:362
>>>>>>>> (sofia/internal/1000 at asterisk ) Running State 
>>>>>>>> Change
>>>>>>>> CS_HANGUP
>>>>>>>> 2011-10-10 17:36:41.428463 [ERR] mod_bwe.c:345
>>>>>>>> 713d5f0c-4341-47ec-b0bf-4b14bd6b4726: Remove 
>>>>>>>> local
>>>>>>>> variable
>>>>>>>> store!
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_media_bug.c:480
>>>>>>>> Removing BUG from sofia/internal/1000 at asterisk 
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:602
>>>>>>>> (sofia/internal/1000 at asterisk ) State HANGUP
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG] 
>>>>>>>> mod_sofia.c:454
>>>>>>>> sofia/internal/1000 at asterisk Overriding SIP cause 
>>>>>>>> 480 with
>>>>>>>> 200
>>>>>>>> from the
>>>>>>>> other leg
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG] 
>>>>>>>> mod_sofia.c:460
>>>>>>>> Channel
>>>>>>>> sofia/internal/1000 at asterisk hanging up, cause:
>>>>>>>> NORMAL_CLEARING
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:47
>>>>>>>> sofia/internal/1000 at asterisk Standard HANGUP, 
>>>>>>>> cause:
>>>>>>>> NORMAL_CLEARING
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:47
>>>>>>>> sofia/internal/sip:1001 at 137.226.198.252 :5060 
>>>>>>>> Standard
>>>>>>>> HANGUP,
>>>>>>>> cause:
>>>>>>>> NORMAL_CLEARING
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:602
>>>>>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) 
>>>>>>>> State
>>>>>>>> HANGUP
>>>>>>>> going to
>>>>>>>> sleep
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:393
>>>>>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) 
>>>>>>>> State
>>>>>>>> Change
>>>>>>>> CS_HANGUP
>>>>>>>> -> CS_REPORTING
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_session.c:1177
>>>>>>>> Send
>>>>>>>> signal sofia/internal/sip:1001 at 137.226.198.252 
>>>>>>>> :5060
>>>>>>>> [BREAK]
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:362
>>>>>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) 
>>>>>>>> Running
>>>>>>>> State
>>>>>>>> Change
>>>>>>>> CS_REPORTING
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:662
>>>>>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) 
>>>>>>>> State
>>>>>>>> REPORTING
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:79
>>>>>>>> sofia/internal/sip:1001 at 137.226.198.252 :5060 
>>>>>>>> Standard
>>>>>>>> REPORTING,
>>>>>>>> cause: NORMAL_CLEARING
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:662
>>>>>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) 
>>>>>>>> State
>>>>>>>> REPORTING going
>>>>>>>> to sleep
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:387
>>>>>>>> (sofia/internal/sip:1001 at 137.226.198.252 :5060) 
>>>>>>>> State
>>>>>>>> Change
>>>>>>>> CS_REPORTING -> CS_DESTROY
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_session.c:1177
>>>>>>>> Send
>>>>>>>> signal sofia/internal/sip:1001 at 137.226.198.252 
>>>>>>>> :5060
>>>>>>>> [BREAK]
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_session.c:1351
>>>>>>>> Session
>>>>>>>> 8 (sofia/internal/sip:1001 at 137.226.198.252 
>>>>>>>> :5060)
>>>>>>>> Locked,
>>>>>>>> Waiting on
>>>>>>>> external entities
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:602
>>>>>>>> (sofia/internal/1000 at asterisk ) State HANGUP going 
>>>>>>>> to
>>>>>>>> sleep
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:393
>>>>>>>> (sofia/internal/1000 at asterisk ) State Change 
>>>>>>>> CS_HANGUP ->
>>>>>>>> CS_REPORTING
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_session.c:1177
>>>>>>>> Send
>>>>>>>> signal sofia/internal/1000 at asterisk [BREAK]
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:362
>>>>>>>> (sofia/internal/1000 at asterisk ) Running State 
>>>>>>>> Change
>>>>>>>> CS_REPORTING
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:662
>>>>>>>> (sofia/internal/1000 at asterisk ) State REPORTING
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:79
>>>>>>>> sofia/internal/1000 at asterisk Standard REPORTING, 
>>>>>>>> cause:
>>>>>>>> NORMAL_CLEARING
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:662
>>>>>>>> (sofia/internal/1000 at asterisk ) State REPORTING 
>>>>>>>> going to
>>>>>>>> sleep
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:387
>>>>>>>> (sofia/internal/1000 at asterisk ) State Change 
>>>>>>>> CS_REPORTING
>>>>>>>> ->
>>>>>>>> CS_DESTROY
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_session.c:1177
>>>>>>>> Send
>>>>>>>> signal sofia/internal/1000 at asterisk [BREAK]
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_session.c:1351
>>>>>>>> Session
>>>>>>>> 7 (sofia/internal/1000 at asterisk ) Locked, Waiting 
>>>>>>>> on
>>>>>>>> external
>>>>>>>> entities
>>>>>>>> 2011-10-10 17:36:41.428463 [NOTICE]
>>>>>>>> switch_core_session.c:1369
>>>>>>>> Session 7 (sofia/internal/1000 at asterisk ) Ended
>>>>>>>> 2011-10-10 17:36:41.428463 [NOTICE]
>>>>>>>> switch_core_session.c:1371
>>>>>>>> Close
>>>>>>>> Channel sofia/internal/1000 at asterisk 
>>>>>>>> [CS_DESTROY]
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:491
>>>>>>>> (sofia/internal/1000 at asterisk ) Callstate Change 
>>>>>>>> HANGUP ->
>>>>>>>> DOWN
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:494
>>>>>>>> (sofia/internal/1000 at asterisk ) Running State 
>>>>>>>> Change
>>>>>>>> CS_DESTROY
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:504
>>>>>>>> (sofia/internal/1000 at asterisk ) State DESTROY
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG] 
>>>>>>>> mod_sofia.c:365
>>>>>>>> sofia/internal/1000 at asterisk SOFIA DESTROY
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:86
>>>>>>>> sofia/internal/1000 at asterisk Standard DESTROY
>>>>>>>> 2011-10-10 17:36:41.428463 [DEBUG]
>>>>>>>> switch_core_state_machine.c:504
>>>>>>>> (sofia/internal/1000 at asterisk ) State DESTROY going 
>>>>>>>> to
>>>>>>>> sleep
>>>>>>>> 
>>>>>>>> Could anyone give me some hints why the remote session is 
>>>>>>>> not
>>>>>>>> ended
>>>>>>>> automatically?
>>>>>>>> 
>>>>>>>> 
>>>>>>>> ----------------urspr?ngliche
>>>>>>>> Nachricht-----------------
>>>>>>>> Von: "Anthony Minessale" 
>>>>>>>> anthony.minessale at gmail.com 
>>>>>>>> An: freeswitch-dev at lists.freeswitch.org 
>>>>>>>> Datum: Mon, 10 Oct 2011 10:17:36 -0500
>>>>>>>> 
>>>>>>>> -------------------------------------------------
>>>>>>>> 
>>>>>>>> 
>>>>>>>>> it calls the callback with type set to
>>>>>>>>> SWITCH_ABC_TYPE_CLOSE
>>>>>>>>> 
>>>>>>>>> On Wed, Oct 5, 2011 at 7:37 PM, Vu Quang Hoa
>>>>>>>>> vutamhoan at gmail.com 
>>>>>>>>> wrote:
>>>>>>>>>> As I tested - FS automatically removes bug when 
>>>>>>>>>> call
>>>>>>>>>> ends.
>>>>>>>>>> I'm trying to figure out if it does the same with
>>>>>>>>>> allocated
>>>>>>>>>> resources
>>>>>>>>>> in
>>>>>>>>>> add_bug routine.
>>>>>>>>>> 
>>>>>>>>>> On Tue, Oct 4, 2011 at 7:03 PM, Han Xin
>>>>>>>>>> xin at ind.rwth-aachen.de 
>>>>>>>>>> wrote:
>>>>>>>>>>> 
>>>>>>>>>>> Hello everyone!
>>>>>>>>>>> 
>>>>>>>>>>> I am new to Freeswitch and try to add media bugs 
>>>>>>>>>>> such
>>>>>>>>>>> as
>>>>>>>>>>> digital
>>>>>>>>>>> signal
>>>>>>>>>>> processing to the read stream and write 
>>>>>>>>>>> stream. But I
>>>>>>>>>>> have
>>>>>>>>>>> some
>>>>>>>>>>> trouble on
>>>>>>>>>>> how to remove the media bug from the session 
>>>>>>>>>>> after
>>>>>>>>>>> the
>>>>>>>>>>> sessions are
>>>>>>>>>>> closed.
>>>>>>>>>>> I tried to look into other existed modules 
>>>>>>>>>>> such as
>>>>>>>>>>> mod_soundtouch
>>>>>>>>>>> and
>>>>>>>>>>> mod_dptools to find a solution, the former
>>>>>>>>>>> mod_soundtouch
>>>>>>>>>>> put
>>>>>>>>>>> the
>>>>>>>>>>> function
>>>>>>>>>>> "switch_core_media_bug_remove()" in the
>>>>>>>>>>> beginning of
>>>>>>>>>>> application
>>>>>>>>>>> run
>>>>>>>>>>> function. The latter mod use a single app 
>>>>>>>>>>> called
>>>>>>>>>>> "remove
>>>>>>>>>>> bugs" and
>>>>>>>>>>> put
>>>>>>>>>>> this
>>>>>>>>>>> app in the dial plan to remove the bugs.
>>>>>>>>>>> 
>>>>>>>>>>> The question is where should I put this
>>>>>>>>>>> "switch_core_media_bug_remove()"
>>>>>>>>>>> function after the sessions are closed.
>>>>>>>>>>> Is there a way to automatically remove a media 
>>>>>>>>>>> bug
>>>>>>>>>>> when a call
>>>>>>>>>>> is
>>>>>>>>>>> ended?
>>>>>>>>>>> Or it is only possible to use a single app to 
>>>>>>>>>>> remove the
>>>>>>>>>>> media
>>>>>>>>>>> bugs as
>>>>>>>>>>> in
>>>>>>>>>>> the mod_dptools?
>>>>>>>>>>> 
>>>>>>>>>>> Thank you all in advance!
>>>>>>>>>>> 
>>>>>>>>>>> --
>>>>>>>>>>> 
>>>>>>>>>>> 
>>>>>>>>>>> 
>>>>>>>>>>> 
>>>>>>>>>>> 
>>>>>>>>>>> 
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>>> 
>>>>>>>>>>> FreeSWITCH-dev mailing list
>>>>>>>>>>> FreeSWITCH-dev at lists.freeswitch.org 
>>>>>>>>>>> 
>>>>>>>>>>> 
>>>>>>>>>>> 
>>>>>>>>>>> 
>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/fre
>>>>>>>>>>> eswi
>>>>>>>>>>> tch-
>>>>>>>>>>> dev
>>>>>>>>>>> 
>>>>>>>>>>> 
>>>>>>>>>>> 
>>>>>>>>>>> 
>>>>>>>>>>> 
>>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/
>>>>>>>>>>> opti
>>>>>>>>>>> ons/
>>>>>>>>>>> free
>>>>>>>>>>> swit
>>>>>>>>>>> ch-dev
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> _______________________________________________
>>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>>> 
>>>>>>>>>> FreeSWITCH-dev mailing list
>>>>>>>>>> FreeSWITCH-dev at lists.freeswitch.org 
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/frees
>>>>>>>>>> witc
>>>>>>>>>> h-de
>>>>>>>>>> v
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/op
>>>>>>>>>> tion
>>>>>>>>>> s/fr
>>>>>>>>>> eesw
>>>>>>>>>> itch
>>>>>>>>>> -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
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>> _______________________________________________
>>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>>> 
>>>>>>>>> FreeSWITCH-dev mailing list
>>>>>>>>> FreeSWITCH-dev at lists.freeswitch.org 
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswi
>>>>>>>>> tch-
>>>>>>>>> dev
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/opti
>>>>>>>>> ons/
>>>>>>>>> free
>>>>>>>>> swit
>>>>>>>>> ch-d
>>>>>>>>> ev
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> 
>>>>>>>> 
>>>>>>>> --
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> _______________________________________________
>>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>>> 
>>>>>>>> FreeSWITCH-dev mailing list
>>>>>>>> FreeSWITCH-dev at lists.freeswitch.org 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitc
>>>>>>>> h-de
>>>>>>>> v
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/option
>>>>>>>> s/fr
>>>>>>>> eesw
>>>>>>>> itch
>>>>>>>> -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
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>>> 
>>>>>>> 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/
>>>>>>> free
>>>>>>> swit
>>>>>>> ch-d
>>>>>>> ev
>>>>>>> http://www.freeswitch.org
>>>>>>> 
>>>>>> 
>>>>>> --
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> _______________________________________________
>>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>>> 
>>>>>> FreeSWITCH-dev mailing list
>>>>>> FreeSWITCH-dev at lists.freeswitch.org 
>>>>>> 
>>>>>> 
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-de
>>>>>> v
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/fr
>>>>>> eesw
>>>>>> itch
>>>>>> -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
>>>>> 
>>>>> _______________________________________________
>>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>>> http://www.cluecon.com 877-7-4ACLUE
>>>>> 
>>>>> 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/free
>>>>> swit
>>>>> ch-d
>>>>> ev
>>>>> http://www.freeswitch.org
>>>>> 
>>>> 
>>>> --
>>>> 
>>>> 
>>>> 
>>>> 
>>>> _______________________________________________
>>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>>> http://www.cluecon.com 877-7-4ACLUE
>>>> 
>>>> 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/freesw
>>>> itch
>>>> -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
>>> 
>>> _______________________________________________
>>> Join us at ClueCon 2011, Aug 9-11, Chicago
>>> http://www.cluecon.com 877-7-4ACLUE
>>> 
>>> 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/freeswit
>>> ch-d
>>> ev
>>> http://www.freeswitch.org
>>> 
>> 
>> --
>> 
>> <mod_bwe_debug.c>_______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>> 
>> 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
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> 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-d
> ev
> http://www.freeswitch.org
> 

-- 

 



From peter.olsson at visionutveckling.se  Mon Oct 24 15:39:42 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 24 Oct 2011 13:39:42 +0200
Subject: [Freeswitch-dev] Build issues on Windows (was: FreeSWITCH-dev
 Digest, Vol 64, Issue 19)
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF177@cooper>
References: <mailman.7718.1319352469.29749.freeswitch-dev@lists.freeswitch.org>,
	<CADLgD89euTxBo-qeVLqBhE-x76NBz8gDBGBpwrG6EFGgVTGAkQ@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF177@cooper>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59F2AEC514@cooper>

Update to this case:

For mod_sofia I've created this Jira issue (with attached patch): FS-3633. Except this I found one more thing, it seems like the first build (at least in VS2008) might fail to build openssl before iksemel, which causes some headers to be missing. In that case, just build openssl project separately and then rebuild the solution (this only needs to be done once, a complete clean/rebuild won't require this after it's been done the first time). Except this everything is working just fine for me.

I suggest you check that you don't have autocrlf set till true in git, this is a common misstake among Windows users.

/Peter



-----Ursprungligt meddelande-----
Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] F?r Peter Olsson
Skickat: den 24 oktober 2011 06:56
Till: freeswitch-dev at lists.freeswitch.org
?mne: Re: [Freeswitch-dev] FreeSWITCH-dev Digest, Vol 64, Issue 19

I build FS on Windows every day, I rarely have problems with it, and if a problem occurs it's mostly because of a change in the code that works on Linux, but not Windows - but these kind of problems are usually fixed later the same day (or i just submit a patch, and it will be applied soon enough).

Last time I did a complete rebuild of FS (with no libs downloaded) I also found some problems with openssl (but this has only happened once), but I just removed the openssl-folder and rebuilt, and everything worked, the rest of the things you mention here I've never heard of before. My guess is you have problems related to antivirus or UAC on your machine, but I will try to do a complete new rebuild today, and see how it looks for me - I'll get back to you after that.

Also, please don't hijack other threads, send a new mail the next time, instead of replying to a digest mail.

/Peter
________________________________________
Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [freeswitch-dev-bounces at lists.freeswitch.org] f&#246;r marco blaisou [blaisounou at gmail.com]
Skickat: den 23 oktober 2011 17:13
Till: freeswitch-dev at lists.freeswitch.org
?mne: Re: [Freeswitch-dev] FreeSWITCH-dev Digest, Vol 64, Issue 19

COMPILING FREESWITCH ON WINDOWS 7 FROM THE GIT
-----------------------------------------------------------------------------------------

Hi guys, just to give you some update from the trenches and give you some update on
compiling FreeSwitch on Windows from the git.(what a journey!!!)

I can tell you that so far, it appeared to be one of the most frustrating experience in
building a system from a git: files are missing or are not loaded properly as they should...

So after a few nights of sctatching my head, here is the status:


Building on Windows from a git repository
-------------------------------------------------------------

After git cloning the repository, open vs2008 solution then:

Stop Comodo or any firewall and virus detection
(if not this screws up download scripts giving: util.vbs(184, 2) msxml3.dll errors)

0) launch build

1) insure all the supposed dowloaded directories are present. (!!!!!!!!!!!!! VITAL !!!!!!!!)
Actually there are some problems:

-> issue with opensssl download
-------------------------------------------------
for a strange reason after autodownload  the untar is not complete, resulting in missing subdirectory
So make sure to do it by hand if necessary

-> issue with CELT download
-------------------------------------------
MUST change prebuilt instructions to :
if not exist "$(ProjectDir)..\celt-0.10.0" cscript /nologo "$(ProjectDir)util.vbs" GetUnzip http://files.freeswitch.org/downloads/libs/celt-0.10.0.tar.gz "$(ProjectDir).."
xcopy "$(ProjectDir)\celt\config.h" "$(ProjectDir)..\celt-0.10.0\libcelt\" /C /D /Y /S /I
xcopy "$(ProjectDir)\celt\float_cast.h" "$(ProjectDir)..\celt-0.10.0\libcelt\" /C /D /Y /S /I


2) if you uses Windows sdk 7.1

edit fake-rfc2533.h and comment out :

#ifdef nodef // *****add this
#ifndef HAVE_STRUCT_ADDRINFO

struct addrinfo {
int    ai_flags;    /* AI_PASSIVE, AI_CANONNAME */
int    ai_family;    /* PF_xxx */
int    ai_socktype;    /* SOCK_xxx */
int    ai_protocol;    /* 0 or IPPROTO_xxx for IPv4 and IPv6 */
size_t    ai_addrlen;    /* length of ai_addr */
char    *ai_canonname;    /* canonical name for hostname */
struct sockaddr *ai_addr;    /* binary address */
struct addrinfo *ai_next;    /* next structure in linked list */
};
#endif /* !HAVE_STRUCT_ADDRINFO */

#endif      // *****add this


3) if issue with folder: sofia-sip
cannot include file sip-extra.h ???????????????????????????????????????

(I'm fed up with this so I just decided to get it from a tar)

4) pb on compile : libsndfile

in sofia_presence.c
    if (mod_sofia_globals.debug_presence > 1) {
        int i;                    // to be added
        for (i = 0; i < argc; i++) {

//----------------------------------------------------------------------------------

That's it for the time being, most of the stuff (but not ALL) now compile
and I'll try to focus on other pending issues.

OBVIOUSLY ANY ADVICE IS MORE THEN WELCOME !!!!

2011/10/23 <freeswitch-dev-request at lists.freeswitch.org<mailto:freeswitch-dev-request at lists.freeswitch.org>>
Send FreeSWITCH-dev mailing list submissions to
       freeswitch-dev at lists.freeswitch.org<mailto:freeswitch-dev at lists.freeswitch.org>

To subscribe or unsubscribe via the World Wide Web, visit
       http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
or, via email, send a message with subject or body 'help' to
       freeswitch-dev-request at lists.freeswitch.org<mailto:freeswitch-dev-request at lists.freeswitch.org>

You can reach the person managing the list at
       freeswitch-dev-owner at lists.freeswitch.org<mailto:freeswitch-dev-owner at lists.freeswitch.org>

When replying, please edit your Subject line so it is more specific
than "Re: Contents of FreeSWITCH-dev digest..."

Today's Topics:

  1. == ISSUE building FreeSwitch on Windows (marco blaisou)
  2. Re: Strange behaviour in SOFIA SIP (Visciano Giovanni)
  3. Re: == ISSUE building FreeSwitch on Windows (Peter Olsson)


---------- Message transf?r? ----------
From: marco blaisou <blaisounou at gmail.com<mailto:blaisounou at gmail.com>>
To: freeswitch-dev at lists.freeswitch.org<mailto:freeswitch-dev at lists.freeswitch.org>
Date: Sat, 22 Oct 2011 16:59:56 +0200
Subject: [Freeswitch-dev] == ISSUE building FreeSwitch on Windows
Hi guys,
For the few days I've been unsuccesfull in building FreeSwith on Windows 7

I did follow the Wiki line per line, but some thing is really going wrong:
> apparently the files in /libs directory are not loaded from the net (ie the util.vbs does seem to load and unpack the libs)
error is : freeswitch\libs\win32\util.vbs(184, 2) msxml3.dll: Le syst?me ne trouve pas la ressource sp?cifi?e (ie system doesn't find specified resource)

> then obviously the compiler starts complaining about missing stuff such as:
c:\users\jc\desktop\voip\freeswitch\freeswitch\src\mod\endpoints\mod_sofia\mod_sofia.h(47) : fatal error C1083: Cannot open include file: 'switch_version.h': No such file or directory

To summarize:
I loaded the FreeSwitch git repository with TortoiseGit

My config is :
Windows 7/64bits
VSC2008 sp1
Windows SDK 7.1

My firewall is desactivated as suggested

I did follow the recommendation to clear some files:
del libs\apr\include\apr.h
del libs\js\config.h
del libs\js\src\jsautocfg.h
del libs\js\nsprpub\pr\include\prcpucfg.h
del libs\iksemel\include\config.h
del libs\xmlrpc-c\xmlrpc_config.h
del libs\libsndfile\src\sfconfig.h
del libs\sofia-sip\libsofia-sip-ua\su\sofia-sip\su_configure.h
del libs\js\nsprpub\pr\include\prcpucfg.h

cd libs\win32\libsndfile
rename config.h sfconfig.h

Could a guru provide some hints on how to solve this???
Thanks




---------- Message transf?r? ----------
From: "Visciano Giovanni" <Giovanni.Visciano at italtel.it<mailto:Giovanni.Visciano at italtel.it>>
To: <freeswitch-dev at lists.freeswitch.org<mailto:freeswitch-dev at lists.freeswitch.org>>
Date: Sat, 22 Oct 2011 20:30:06 +0200
Subject: Re: [Freeswitch-dev] Strange behaviour in SOFIA SIP
Now I understand.
If both legs use the same codec, the codec ("fake") will be used in a pass-through way,
so no additional transcoding is applied, as in PROXY PASS-THROUGH condition.

So let me reason in a default media scenario (no proxy-media).

Now, as I explained in the first mail, sofia build "ep_codec_string" (SWITCH_
ORIGINATOR_CODEC_VARIABLE) as the intersection between the codec list received
in SDP offer and the codec loaded in the core.
So if the offered SDP codec list is NOT a subset of the loaded core codecs the other
legs will miss some codecs proposed by the calling endpoint.
That said, if I want to be sure every codecs my endpoint support won't be filtered I
have to register a "fake" pass-through codec for each one.

... you did not answer to my last question :).
I haven't found any other endpoint module the can interwork in PROXY mode with
sofia (even h323 doesn't support this). My question: is there a standard/
recommended way to exchange capability between endpoint of different nature in FS
proxy mode? Maybe using SDP o/a as done in sofia sip?

Thank you again for you help.
Giovanni


-----Original Message-----
From: freeswitch-dev-bounces at lists.freeswitch.org<mailto:freeswitch-dev-bounces at lists.freeswitch.org> on behalf of Anthony Minessale
Sent: Sat 10/22/2011 4:48 PM
To: freeswitch-dev at lists.freeswitch.org<mailto:freeswitch-dev at lists.freeswitch.org>
Subject: Re: [Freeswitch-dev] Strange behaviour in SOFIA SIP

that's what I was trying to explain.
PROXY mode does not let you manipulate anything, it's pure
pass-through negotiation between the endpoints.

All the stuff with ep_codec_string are relevant only when you are not
trying to use proxy mode.
This is a common mistake people make assuming not using proxy mode
means somehow doing more work but it's not the case, any time both
legs of a call use the same codec it's equally efficient to what you
end up with in proxy mode.  proxy mode simply means take the inbound
SDP and offer it AS IS to the far end and vice-versa and use a fake
codec in the middle that is agnostic.



On Fri, Oct 21, 2011 at 10:36 AM, Visciano Giovanni
<Giovanni.Visciano at italtel.it<mailto:Giovanni.Visciano at italtel.it>> wrote:
> Hello Anthony.
> Maybe I want not so clear, let me explain.
>
> I do not want to apply any codec manipulation, I'm working on a prototype endpoint module that has "proxy media capabilty" and wants to interoperate with other endpoints.
> What I saw is that in a sofia vs sofia call (proxy mode+late negotiation) they exchange through FS calling/called endpoint capability looking at SDP session carried by channel variables, so internally they use SDP O/A scheme.
> I thought that "ep_codec_string"->SWITCH_ORIGINATOR_CODEC_VARIABLE could be used to exchange capability without using a protocol specific format (SDP).
> If this is not the case, what's the right way to negotiate capability between endpoints in proxy mode?
>
> Than you.
> Giovanni
>
> PS: I did not find any existing endpoint except sofia that support this. Am I right?
>
> -----Original Message-----
> From: freeswitch-dev-bounces at lists.freeswitch.org<mailto:freeswitch-dev-bounces at lists.freeswitch.org> on behalf of Anthony Minessale
> Sent: Fri 10/21/2011 5:06 PM
> To: freeswitch-dev at lists.freeswitch.org<mailto:freeswitch-dev at lists.freeswitch.org>
> Subject: Re: [Freeswitch-dev] Strange behaviour in SOFIA SIP
>
> any codec manipulation techniques and proxy media mode are not
> compatible.  Proxy media mode means stay completely out of the codec
> negotiation process.
>
> On Thu, Oct 20, 2011 at 3:11 PM, Visciano Giovanni
> <Giovanni.Visciano at italtel.it<mailto:Giovanni.Visciano at italtel.it>> wrote:
>> Preconditions on sip profile
>> ----------------------------
>> inbound-codec-prefs NOT set
>> outbound-codec-prefs NOT set
>> inbound-proxy-media=TRUE
>> inbound-bypass-media=FALSE
>> inbound-late-negotiation=TRUE + inherit_codec=TRUE
>>
>> TEST scenario
>> -------------
>> SIP phone send INVITE with SDP to FS
>>
>> INVITE
>> SDP[codelist=PCMA, PCMU, G722, G729] --> FS
>>
>> FS has just the CORE_PC_MODULE [PCMA, PCMU, PASS-THR, VIDEO-PASS-THR, L16]
>>
>> The "ep_codec_string" channel variable is loaded with [PCMA, PCMU]
>>
>> EXPECTED
>> --------
>> In "proxy-media + late-negotiation" scenario:
>> ep_codec_string=[PCMA,PCMU,G722,G729]
>>
>> PROBLEM
>> -------
>> According to the documentation
>> (http://wiki.freeswitch.org/wiki/Codec_negotiation):
>>
>> "ep_codec_string: This variable is only available if late negotiation is
>> enabled on the profile. It's a readable string containing all the codecs
>> proposed by the calling endpoint."
>> and
>> "inherit_codec=true (only applicable when late-negotiation is enabled) will
>> take the codec negotiated when the B leg answers and pass it to the A leg so
>> it also uses the same codec (if the A leg supports it); otherwise, it will
>> use whatever it can from its own list"
>>
>> The variable ep_codec_string doesn't contain ALL the codecs proposed by the
>> calling endpoint. It contains the intersection between the codecs proposed
>> by the calling and the codecs loaded in FS.
>> In proxy-media / no-media configuration the called endpoint doesn't have the
>> calling codec list so he can't negotiate using all the capabilities
>> available.
>> (I'm working on a test endpoint module that should work in proxy and
>> no-media configuration ... mod_sofia vs mod_sofia works because they look at
>> the remote sdp variable)
>>
>> Source reference - sofia_glue.c->sofia_glue_set_r_sdp_codec_string()
>>
>>
>> Regards,
>> Giovanni
>>
>>
>>
>> Internet Email Confidentiality Footer
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>
>
>
> --
> Anthony Minessale II
>
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> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
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>
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>
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--
Anthony Minessale II

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ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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_______________________________________________
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---------- Message transf?r? ----------
From: Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>>
To: "freeswitch-dev at lists.freeswitch.org<mailto:freeswitch-dev at lists.freeswitch.org>" <freeswitch-dev at lists.freeswitch.org<mailto:freeswitch-dev at lists.freeswitch.org>>
Date: Sun, 23 Oct 2011 08:42:42 +0200
Subject: Re: [Freeswitch-dev] == ISSUE building FreeSwitch on Windows
I believe that you might be missing some system dll's on your machine.

msxml3.dll is something that should exist on your computer, and I think the download script is depending on it.

/Peter
________________________________________
Fr?n: freeswitch-dev-bounces at lists.freeswitch.org<mailto:freeswitch-dev-bounces at lists.freeswitch.org> [freeswitch-dev-bounces at lists.freeswitch.org<mailto:freeswitch-dev-bounces at lists.freeswitch.org>] f&#246;r marco blaisou [blaisounou at gmail.com<mailto:blaisounou at gmail.com>]
Skickat: den 22 oktober 2011 16:59
Till: freeswitch-dev at lists.freeswitch.org<mailto:freeswitch-dev at lists.freeswitch.org>
?mne: [Freeswitch-dev] == ISSUE building FreeSwitch on Windows

Hi guys,
For the few days I've been unsuccesfull in building FreeSwith on Windows 7

I did follow the Wiki line per line, but some thing is really going wrong:
> apparently the files in /libs directory are not loaded from the net (ie the util.vbs does seem to load and unpack the libs)
error is : freeswitch\libs\win32\util.vbs(184, 2) msxml3.dll: Le syst?me ne trouve pas la ressource sp?cifi?e (ie system doesn't find specified resource)

> then obviously the compiler starts complaining about missing stuff such as:
c:\users\jc\desktop\voip\freeswitch\freeswitch\src\mod\endpoints\mod_sofia\mod_sofia.h(47) : fatal error C1083: Cannot open include file: 'switch_version.h': No such file or directory

To summarize:
I loaded the FreeSwitch git repository with TortoiseGit

My config is :
Windows 7/64bits
VSC2008 sp1
Windows SDK 7.1

My firewall is desactivated as suggested

I did follow the recommendation to clear some files:
del libs\apr\include\apr.h
del libs\js\config.h
del libs\js\src\jsautocfg.h
del libs\js\nsprpub\pr\include\prcpucfg.h
del libs\iksemel\include\config.h
del libs\xmlrpc-c\xmlrpc_config.h
del libs\libsndfile\src\sfconfig.h
del libs\sofia-sip\libsofia-sip-ua\su\sofia-sip\su_configure.h
del libs\js\nsprpub\pr\include\prcpucfg.h

cd libs\win32\libsndfile
rename config.h sfconfig.h

Could a guru provide some hints on how to solve this???
Thanks






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!DSPAM:4ea4ef5432761992943852!



From cbeeton at avaya.com  Mon Oct 24 17:55:36 2011
From: cbeeton at avaya.com (Beeton, Carolyn (Carolyn))
Date: Mon, 24 Oct 2011 09:55:36 -0400
Subject: [Freeswitch-dev] FS-2877 patch
Message-ID: <E151F470685C15418AEA609E825A171022BE8A9494@DC-US1MBEX4.global.avaya.com>

Anthony,

I see that the patch from FS-2877 has been committed (thanks!  I hope it helps to address the BLF issues others have seen).  I did have a comment on that issue that there was a typo in the patch which I don't think was corrected in the commit.  The typo will cause the database to be dropped and rebuilt on every run.  This would fix it:

diff --git a/src/mod/endpoints/mod_sofia/sofia_glue.c b/src/mod/endpoints/mod_sofia/sofia_glue.c
index e6076e8..a6de2f5 100644
--- a/src/mod/endpoints/mod_sofia/sofia_glue.c
+++ b/src/mod/endpoints/mod_sofia/sofia_glue.c
@@ -5974,7 +5974,7 @@ int sofia_glue_init_sql(sofia_profile_t *profile)
  switch_cache_db_test_reactive(dbh, test_sql, "DROP TABLE sip_subscriptions", sub_sql);

  free(test_sql);
- test_sql = switch_mprintf("delete from sip_dialogs where hostname='%q' and expires <> -9999 or rpid='' or from_tag=''", mod_sofia_globals.hostname);
+ test_sql = switch_mprintf("delete from sip_dialogs where hostname='%q' and expires <> -9999 or rpid='' or sip_from_tag=''", mod_sofia_globals.hostname);


  switch_cache_db_test_reactive(dbh, test_sql, "DROP TABLE sip_dialogs", dialog_sql);


Carolyn
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From anthony.minessale at gmail.com  Mon Oct 24 18:40:26 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 24 Oct 2011 09:40:26 -0500
Subject: [Freeswitch-dev] FS-2877 patch
In-Reply-To: <E151F470685C15418AEA609E825A171022BE8A9494@DC-US1MBEX4.global.avaya.com>
References: <E151F470685C15418AEA609E825A171022BE8A9494@DC-US1MBEX4.global.avaya.com>
Message-ID: <CAKbxfG8sZe78Nig1qrbc_=3BSjDneBqtXGGGV-1O_e8KLNFRCA@mail.gmail.com>

got it, thx

On Mon, Oct 24, 2011 at 8:55 AM, Beeton, Carolyn (Carolyn) <
cbeeton at avaya.com> wrote:

> **
> Anthony,
>
> I see that the patch from FS-2877 has been committed (thanks!  I hope it
> helps to address the BLF issues others have seen).  I did have a comment on
> that issue that there was a typo in the patch which I don't think was
> corrected in the commit.  The typo will cause the database to be dropped and
> rebuilt on every run.  This would fix it:
>
> diff --git a/src/mod/endpoints/mod_sofia/sofia_glue.c
> b/src/mod/endpoints/mod_sofia/sofia_glue.c
> index e6076e8..a6de2f5 100644
> --- a/src/mod/endpoints/mod_sofia/sofia_glue.c
> +++ b/src/mod/endpoints/mod_sofia/sofia_glue.c
> @@ -5974,7 +5974,7 @@ int sofia_glue_init_sql(sofia_profile_t *profile)
>   switch_cache_db_test_reactive(dbh, test_sql, "DROP TABLE
> sip_subscriptions", sub_sql);
>
>   free(test_sql);
> - test_sql = switch_mprintf("delete from sip_dialogs where hostname='%q'
> and expires <> -9999 or rpid='' or from_tag=''",
> mod_sofia_globals.hostname);
> + test_sql = switch_mprintf("delete from sip_dialogs where hostname='%q'
> and expires <> -9999 or rpid='' or sip_from_tag=''",
> mod_sofia_globals.hostname);
>
>
>   switch_cache_db_test_reactive(dbh, test_sql, "DROP TABLE sip_dialogs",
> dialog_sql);
>
>
> Carolyn
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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
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From jcherukuri_necc at yahoo.com  Mon Oct 24 18:44:34 2011
From: jcherukuri_necc at yahoo.com (Jyotshna Cherukuri)
Date: Mon, 24 Oct 2011 07:44:34 -0700 (PDT)
Subject: [Freeswitch-dev] Music on Hold gets choppy and cuts off
	intermittently when VAD is enabled on Freeswitch
Message-ID: <1319467474.90450.YahooMailNeo@web122501.mail.ne1.yahoo.com>



Hi,

I have 2 user agent SIP profiles set up on my Freeswitch server(one profile uses PCMU as its codec-prefs and other uses SILK at 8000Hz) and I enabled VAD on PCMU sip profile.

When I make a phone call which involves SILK-PCMU transcoding and if I put PCMU user ON HOLD, I am seeing an issue with MOH being cut off . It seems to quit after playing for a while , breaking through only transiently. It doesn't seem good to the party listening to MOH. This doesn't seem to happen when VAD is disabled though. I had to enable VAD as I was constantly hearing hissing background noise during conversation if VAD was not enabled.

Are there any settings I can adjust in Freeswitch to disable VAD when wav files are bing played in general. Any help is greatly appreciated

Thanks in advance
Regards
Jyotshna
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From steveayre at gmail.com  Mon Oct 24 19:30:54 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 24 Oct 2011 16:30:54 +0100
Subject: [Freeswitch-dev] Strange behaviour in SOFIA SIP
In-Reply-To: <AF4EB29748E178458A0067661112724B02B48F1A@BESONE.corp.dom>
References: <AF4EB29748E178458A0067661112724B04EC6B1B@BESONE.corp.dom>
	<AF4EB29748E178458A0067661112724B02B48F17@BESONE.corp.dom>
	<CAKbxfG-U3BU7mha9zeN-_cWpXjbhwMFBAkuva=rrg_dn8Nvz+w@mail.gmail.com>
	<AF4EB29748E178458A0067661112724B02B48F19@BESONE.corp.dom>
	<CAKbxfG_Y4Q50kFryKEonNAJ9f1pGUcTwE6haH1GjcVAkUYwsQw@mail.gmail.com>
	<AF4EB29748E178458A0067661112724B02B48F1A@BESONE.corp.dom>
Message-ID: <08ED4278-B356-4A21-916E-F5AB3B585D84@gmail.com>

"My question: is there a standard/
recommended way to exchange capability between endpoint of different nature in FS
proxy mode? Maybe using SDP o/a as done in sofia sip?"

Proxy media copies SDP straight through to remove FS from the negotiation process. No other protocols use SDP so I'm not sure how you could avoid the negotiation step with other protocols.

Steve on iPhone

On 22 Oct 2011, at 19:30, "Visciano Giovanni" <Giovanni.Visciano at italtel.it> wrote:

> My question: is there a standard/
> recommended way to exchange capability between endpoint of different nature in FS
> proxy mode? Maybe using SDP o/a as done in sofia sip?


From anthony.minessale at gmail.com  Mon Oct 24 19:58:36 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 24 Oct 2011 10:58:36 -0500
Subject: [Freeswitch-dev] Strange behaviour in SOFIA SIP
In-Reply-To: <AF4EB29748E178458A0067661112724B02B48F1A@BESONE.corp.dom>
References: <AF4EB29748E178458A0067661112724B04EC6B1B@BESONE.corp.dom>
	<AF4EB29748E178458A0067661112724B02B48F17@BESONE.corp.dom>
	<CAKbxfG-U3BU7mha9zeN-_cWpXjbhwMFBAkuva=rrg_dn8Nvz+w@mail.gmail.com>
	<AF4EB29748E178458A0067661112724B02B48F19@BESONE.corp.dom>
	<CAKbxfG_Y4Q50kFryKEonNAJ9f1pGUcTwE6haH1GjcVAkUYwsQw@mail.gmail.com>
	<AF4EB29748E178458A0067661112724B02B48F1A@BESONE.corp.dom>
Message-ID: <CAKbxfG-gLE=uEUYz7iuAVgZFZSiBc2T0ZEXZs+4R3XeJoYAvnA@mail.gmail.com>

everything you were trying to do originally is valid as long as you do not
use proxy mode.
Proxy mode has nearly no benefit at this point.  It was originally designed
for t38 passthru which has now been fully implemented.

You'll struggle to find a codec that is not already supported in FS, very
few are stubs most are full functional.



On Sat, Oct 22, 2011 at 1:30 PM, Visciano Giovanni <
Giovanni.Visciano at italtel.it> wrote:

> Now I understand.
> If both legs use the same codec, the codec ("fake") will be used in a
> pass-through way,
> so no additional transcoding is applied, as in PROXY PASS-THROUGH
> condition.
>
> So let me reason in a default media scenario (no proxy-media).
>
> Now, as I explained in the first mail, sofia build "ep_codec_string"
> (SWITCH_
> ORIGINATOR_CODEC_VARIABLE) as the intersection between the codec list
> received
> in SDP offer and the codec loaded in the core.
> So if the offered SDP codec list is NOT a subset of the loaded core codecs
> the other
> legs will miss some codecs proposed by the calling endpoint.
> That said, if I want to be sure every codecs my endpoint support won't be
> filtered I
> have to register a "fake" pass-through codec for each one.
>
> ... you did not answer to my last question :).
> I haven't found any other endpoint module the can interwork in PROXY mode
> with
> sofia (even h323 doesn't support this). My question: is there a standard/
> recommended way to exchange capability between endpoint of different nature
> in FS
> proxy mode? Maybe using SDP o/a as done in sofia sip?
>
> Thank you again for you help.
> Giovanni
>
>
> -----Original Message-----
> From: freeswitch-dev-bounces at lists.freeswitch.org on behalf of Anthony
> Minessale
> Sent: Sat 10/22/2011 4:48 PM
> To: freeswitch-dev at lists.freeswitch.org
> Subject: Re: [Freeswitch-dev] Strange behaviour in SOFIA SIP
>
> that's what I was trying to explain.
> PROXY mode does not let you manipulate anything, it's pure
> pass-through negotiation between the endpoints.
>
> All the stuff with ep_codec_string are relevant only when you are not
> trying to use proxy mode.
> This is a common mistake people make assuming not using proxy mode
> means somehow doing more work but it's not the case, any time both
> legs of a call use the same codec it's equally efficient to what you
> end up with in proxy mode.  proxy mode simply means take the inbound
> SDP and offer it AS IS to the far end and vice-versa and use a fake
> codec in the middle that is agnostic.
>
>
>
> On Fri, Oct 21, 2011 at 10:36 AM, Visciano Giovanni
> <Giovanni.Visciano at italtel.it> wrote:
> > Hello Anthony.
> > Maybe I want not so clear, let me explain.
> >
> > I do not want to apply any codec manipulation, I'm working on a prototype
> endpoint module that has "proxy media capabilty" and wants to interoperate
> with other endpoints.
> > What I saw is that in a sofia vs sofia call (proxy mode+late negotiation)
> they exchange through FS calling/called endpoint capability looking at SDP
> session carried by channel variables, so internally they use SDP O/A scheme.
> > I thought that "ep_codec_string"->SWITCH_ORIGINATOR_CODEC_VARIABLE could
> be used to exchange capability without using a protocol specific format
> (SDP).
> > If this is not the case, what's the right way to negotiate capability
> between endpoints in proxy mode?
> >
> > Than you.
> > Giovanni
> >
> > PS: I did not find any existing endpoint except sofia that support this.
> Am I right?
> >
> > -----Original Message-----
> > From: freeswitch-dev-bounces at lists.freeswitch.org on behalf of Anthony
> Minessale
> > Sent: Fri 10/21/2011 5:06 PM
> > To: freeswitch-dev at lists.freeswitch.org
> > Subject: Re: [Freeswitch-dev] Strange behaviour in SOFIA SIP
> >
> > any codec manipulation techniques and proxy media mode are not
> > compatible.  Proxy media mode means stay completely out of the codec
> > negotiation process.
> >
> > On Thu, Oct 20, 2011 at 3:11 PM, Visciano Giovanni
> > <Giovanni.Visciano at italtel.it> wrote:
> >> Preconditions on sip profile
> >> ----------------------------
> >> inbound-codec-prefs NOT set
> >> outbound-codec-prefs NOT set
> >> inbound-proxy-media=TRUE
> >> inbound-bypass-media=FALSE
> >> inbound-late-negotiation=TRUE + inherit_codec=TRUE
> >>
> >> TEST scenario
> >> -------------
> >> SIP phone send INVITE with SDP to FS
> >>
> >> INVITE
> >> SDP[codelist=PCMA, PCMU, G722, G729] --> FS
> >>
> >> FS has just the CORE_PC_MODULE [PCMA, PCMU, PASS-THR, VIDEO-PASS-THR,
> L16]
> >>
> >> The "ep_codec_string" channel variable is loaded with [PCMA, PCMU]
> >>
> >> EXPECTED
> >> --------
> >> In "proxy-media + late-negotiation" scenario:
> >> ep_codec_string=[PCMA,PCMU,G722,G729]
> >>
> >> PROBLEM
> >> -------
> >> According to the documentation
> >> (http://wiki.freeswitch.org/wiki/Codec_negotiation):
> >>
> >> "ep_codec_string: This variable is only available if late negotiation is
> >> enabled on the profile. It's a readable string containing all the codecs
> >> proposed by the calling endpoint."
> >> and
> >> "inherit_codec=true (only applicable when late-negotiation is enabled)
> will
> >> take the codec negotiated when the B leg answers and pass it to the A
> leg so
> >> it also uses the same codec (if the A leg supports it); otherwise, it
> will
> >> use whatever it can from its own list"
> >>
> >> The variable ep_codec_string doesn't contain ALL the codecs proposed by
> the
> >> calling endpoint. It contains the intersection between the codecs
> proposed
> >> by the calling and the codecs loaded in FS.
> >> In proxy-media / no-media configuration the called endpoint doesn't have
> the
> >> calling codec list so he can't negotiate using all the capabilities
> >> available.
> >> (I'm working on a test endpoint module that should work in proxy and
> >> no-media configuration ... mod_sofia vs mod_sofia works because they
> look at
> >> the remote sdp variable)
> >>
> >> Source reference - sofia_glue.c->sofia_glue_set_r_sdp_codec_string()
> >>
> >>
> >> Regards,
> >> Giovanni
> >>
> >>
> >>
> >> Internet Email Confidentiality Footer
> >>
> >>
> ********************************************************************************************************************************************
> >>
> >> La presente comunicazione, con le informazioni in essa contenute e ogni
> >> documento o file allegato, e' rivolta unicamente alla/e persona/e cui e'
> >> indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non
> siete
> >> i destinatari/autorizzati siete avvisati che qualsiasi azione, copia,
> >> comunicazione, divulgazione o simili basate sul contenuto di tali
> >> informazioni e' vietata e potrebbe essere contro la legge (art. 616
> C.P.,
> >> D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali).
> Se
> >> avete ricevuto questa comunicazione per errore, vi preghiamo di darne
> >> immediata notizia al mittente e di distruggere il messaggio originale e
> ogni
> >> file allegato senza farne copia alcuna o riprodurne in alcun modo il
> >> contenuto.
> >>
> >> This e-mail and its attachments are intended for the addressee(s) only
> and
> >> are confidential and/or may contain legally privileged information. If
> you
> >> have received this message by mistake or are not one of the addressees
> >> above, you may take no action based on it, and you may not copy or show
> it
> >> to anyone; please reply to this e-mail and point out the error which has
> >> occurred.
> >>
> ********************************************************************************************************************************************
> >>
> >> _______________________________________________
> >> Join us at ClueCon 2011, Aug 9-11, Chicago
> >> http://www.cluecon.com 877-7-4ACLUE
> >>
> >> 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
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > 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
> >
> >
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > 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
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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
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From anthony.minessale at gmail.com  Mon Oct 24 20:11:24 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 24 Oct 2011 11:11:24 -0500
Subject: [Freeswitch-dev] Music on Hold gets choppy and cuts off
 intermittently when VAD is enabled on Freeswitch
In-Reply-To: <1319467474.90450.YahooMailNeo@web122501.mail.ne1.yahoo.com>
References: <1319467474.90450.YahooMailNeo@web122501.mail.ne1.yahoo.com>
Message-ID: <CAKbxfG_q5yJzj-bkCet5XteAsVqJUZRzzEGPY+=VFa3WUe_L=g@mail.gmail.com>

using that server side vad is not really very efficient, I'm sure you could
find a way to flip it on and off but it would take coding.


On Mon, Oct 24, 2011 at 9:44 AM, Jyotshna Cherukuri <
jcherukuri_necc at yahoo.com> wrote:

> Hi,
>
> I have 2 user agent SIP profiles set up on my Freeswitch server(one profile
> uses PCMU as its codec-prefs and other uses SILK at 8000Hz) and I enabled VAD
> on PCMU sip profile.
>
> When I make a phone call which involves SILK-PCMU transcoding and if I put
> PCMU user ON HOLD, I am seeing an issue with MOH being cut off . It seems to
> quit after playing for a while , breaking through only transiently. It
> doesn't seem good to the party listening to MOH. This doesn't seem to happen
> when VAD is disabled though. I had to enable VAD as I was constantly hearing
> hissing background noise during conversation if VAD was not enabled.
>
> Are there any settings I can adjust in Freeswitch to disable VAD when wav
> files are bing played in general. Any help is greatly appreciated
>
> Thanks in advance
> Regards
> Jyotshna
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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
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From Heimo.STIEG at nextiraone.eu  Mon Oct 24 12:44:28 2011
From: Heimo.STIEG at nextiraone.eu (STIEG Heimo)
Date: Mon, 24 Oct 2011 10:44:28 +0200
Subject: [Freeswitch-dev] [patch] Better Python Debug Output
Message-ID: <F31A983FE6A3A846BE1CD384C41A285903DDA15D85@UCMAI002.uc.nextiraone-eu.ned>

Hello,

I've modified the mod_python module to create meaninfull Error Messages:

Instead of this:
2011-10-19 08:28:29.356605 [ERR] mod_python.c:224 Error importing module

We have now this:
2011-10-19 08:28:29.356605 [ERR] mod_python.c:224 Error importing module
2011-10-19 08:28:29.356605 [ERR] mod_python.c:156 Python Error in script nxo_transferifgroup: <type 'exceptions.AttributeError'>: 'module' object has no attribute 'argv'
Exception: None

Traceback:
File: /usr/local/freeswitch/scripts/nxo_transferifgroup.py
Line: 6

Can someone import the patch to the trunk please?
http://jira.freeswitch.org/browse/FS-3615

Regards
Heimo Stieg
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From blaisounou at gmail.com  Mon Oct 24 14:30:37 2011
From: blaisounou at gmail.com (marco blaisou)
Date: Mon, 24 Oct 2011 12:30:37 +0200
Subject: [Freeswitch-dev] Building FS on Windows7/64
Message-ID: <CADLgD8-Ucu8sJRCf2qRtE=-NTqVsWUwvJ2S5CN5oSo6JzqLP9g@mail.gmail.com>

Ok, after digging once more into the problem I decided to use a *Windows
7/32* machine.
The built worked effortless!! A very nice run.

Back to a *Windows 7/64*, I'm still having troubles.
Some of the issue seems related to gawk while building libsofia_sip
And it seems that once the issue occurs, you're stuck with the built....

In any case please note the following remarks:
*

1) if you uses Windows sdk 7.1*

edit fake-rfc2533.h and comment out :

#ifdef nodef // *****add this
#ifndef HAVE_STRUCT_ADDRINFO

struct addrinfo {
int    ai_flags;    /* AI_PASSIVE, AI_CANONNAME */
int    ai_family;    /* PF_xxx */
int    ai_socktype;    /* SOCK_xxx */
int    ai_protocol;    /* 0 or IPPROTO_xxx for IPv4 and IPv6 */
size_t    ai_addrlen;    /* length of ai_addr */
char    *ai_canonname;    /* canonical name for hostname */
struct sockaddr *ai_addr;    /* binary address */
struct addrinfo *ai_next;    /* next structure in linked list */
};
#endif /* !HAVE_STRUCT_ADDRINFO */

#endif      // *****add this

*2) in sofia_presence.c*
   if (mod_sofia_globals.debug_presence > 1) {
       int i;                    // to be added
       for (i = 0; i < argc; i++) {
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From blaisounou at gmail.com  Mon Oct 24 18:41:58 2011
From: blaisounou at gmail.com (marco blaisou)
Date: Mon, 24 Oct 2011 16:41:58 +0200
Subject: [Freeswitch-dev] building on windows 7/64 - follow up
Message-ID: <CADLgD88yRPsy=9LwW=+xztnp3bRP8zvBKH6gu_TCropxSAtrDg@mail.gmail.com>

As a follow up of previous post, the issue is confirmed to be located at the
prebuilt phase of libsofia_sip_ua_static lib
ie at

*if not exist "$(ProjectDir)..\..\sofia-sip\win32\gawk.exe" cscript /nologo
"$(ProjectDir)..\util.vbs" Get
http://files.freeswitch.org/downloads/win32/gawk.exe"$(ProjectDir)..\..\sofia-sip\win32\"
cd /D "$(ProjectDir)..\..\sofia-sip\win32\"
set AWK="$(ProjectDir)..\..\sofia-sip\win32\gawk.exe"
"autogen.cmd"*

What seems to happen is that after gawk.exe is loaded, the prebuilt is
trying to go on while concurrently the anti-virus/protection is checking
the gawk.exe -> this creates a mess and concludes with a failure and then
missing headers and so on.

*What I've done is simply to preload gawk.exe in the target directory,
declared it as safe to the AV/Firewall and only then launch the VS built.*

So as a suggestion, loading gawk ahead (ie at thbeginning of the build)
could be a way

btw. The previous discussed recommendations (changes) related to SDK 7.1 are
valid
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From peter.olsson at visionutveckling.se  Tue Oct 25 00:01:27 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 24 Oct 2011 22:01:27 +0200
Subject: [Freeswitch-dev] Building FS on Windows7/64
In-Reply-To: <CADLgD8-Ucu8sJRCf2qRtE=-NTqVsWUwvJ2S5CN5oSo6JzqLP9g@mail.gmail.com>
References: <CADLgD8-Ucu8sJRCf2qRtE=-NTqVsWUwvJ2S5CN5oSo6JzqLP9g@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59F1FBF17A@cooper>

Thanks for the update. Issue #2 is resolved in latest git. For the other one, please file a Jira, so it can be fixed properly.

/Peter
________________________________________
Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [freeswitch-dev-bounces at lists.freeswitch.org] f&#246;r marco blaisou [blaisounou at gmail.com]
Skickat: den 24 oktober 2011 12:30
Till: freeswitch-dev at lists.freeswitch.org
?mne: [Freeswitch-dev] Building FS on Windows7/64

Ok, after digging once more into the problem I decided to use a Windows 7/32 machine.
The built worked effortless!! A very nice run.

Back to a Windows 7/64, I'm still having troubles.
Some of the issue seems related to gawk while building libsofia_sip
And it seems that once the issue occurs, you're stuck with the built....

In any case please note the following remarks:


1) if you uses Windows sdk 7.1

edit fake-rfc2533.h and comment out :

#ifdef nodef // *****add this
#ifndef HAVE_STRUCT_ADDRINFO

struct addrinfo {
int    ai_flags;    /* AI_PASSIVE, AI_CANONNAME */
int    ai_family;    /* PF_xxx */
int    ai_socktype;    /* SOCK_xxx */
int    ai_protocol;    /* 0 or IPPROTO_xxx for IPv4 and IPv6 */
size_t    ai_addrlen;    /* length of ai_addr */
char    *ai_canonname;    /* canonical name for hostname */
struct sockaddr *ai_addr;    /* binary address */
struct addrinfo *ai_next;    /* next structure in linked list */
};
#endif /* !HAVE_STRUCT_ADDRINFO */

#endif      // *****add this

2) in sofia_presence.c
   if (mod_sofia_globals.debug_presence > 1) {
       int i;                    // to be added
       for (i = 0; i < argc; i++) {
!DSPAM:4ea5c15a32761580114226!


From xin at ind.rwth-aachen.de  Tue Oct 25 12:52:36 2011
From: xin at ind.rwth-aachen.de (Han Xin)
Date: Tue, 25 Oct 2011 10:52:36 +0200
Subject: [Freeswitch-dev] choppy sound with .wav file played in the
	background
Message-ID: <ed7de0668f479d709574c9724799f942@gw.ind.rwth-aachen.de>

Hi all, I have another question to ask again :P

When I use:

 switch_ivr_play_file(session, NULL, file_path, NULL);

to play a file such as "misc/8000/sorry.wav" directly, the sound is so choppy.

Am I doing the background sound playing the right way?

Han
--





From peter.olsson at visionutveckling.se  Tue Oct 25 13:00:53 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 25 Oct 2011 11:00:53 +0200
Subject: [Freeswitch-dev] choppy sound with .wav file played in
	the	background
In-Reply-To: <ed7de0668f479d709574c9724799f942@gw.ind.rwth-aachen.de>
References: <ed7de0668f479d709574c9724799f942@gw.ind.rwth-aachen.de>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59F2AEC8D0@cooper>

How do you mean when you say "in background"? Are ju running this command from another thread then the channel's own thread?

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] F?r Han Xin
Skickat: den 25 oktober 2011 10:53
Till: freeswitch-dev at lists.freeswitch.org
?mne: [Freeswitch-dev] choppy sound with .wav file played in the background

Hi all, I have another question to ask again :P

When I use:

 switch_ivr_play_file(session, NULL, file_path, NULL);

to play a file such as "misc/8000/sorry.wav" directly, the sound is so choppy.

Am I doing the background sound playing the right way?

Han
--




_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

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:4ea6780b32761207110701!



From xin at ind.rwth-aachen.de  Tue Oct 25 13:24:06 2011
From: xin at ind.rwth-aachen.de (Han Xin)
Date: Tue, 25 Oct 2011 11:24:06 +0200
Subject: [Freeswitch-dev] choppy sound with .wav file played
	inthebackground
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59F2AEC8D0@cooper>
Message-ID: <ed51f8e938d7265940f67abb91017385@gw.ind.rwth-aachen.de>

 hmm, just one call with two sessions, one is the local session, the other is the remote/partner session.

I want to play a indication sound file when I press a button on each side, while the call is ongoing. That's what I mean with
the background. sorry for the vague description.

Regards,
Han
 
----------------original message-----------------
From: "Peter Olsson" 
To: "'freeswitch-dev at lists.freeswitch.org '" freeswitch-dev at lists.freeswitch.org 
Date: Tue, 25 Oct 2011 11:00:53 +0200
-------------------------------------------------
 
 
> How do you mean when you say "in background"? Are ju running this command from 
> another thread then the channel's own thread?
> 
> /Peter
> 
> 
> -----Ursprungligt meddelande-----
> Fr?n: freeswitch-dev-bounces at lists.freeswitch.org 
> [mailto:freeswitch-dev-bounces at lists.freeswitch.org ] F?r Han Xin
> Skickat: den 25 oktober 2011 10:53
> Till: freeswitch-dev at lists.freeswitch.org 
> ?mne: [Freeswitch-dev] choppy sound with .wav file played in the background
> 
> Hi all, I have another question to ask again :P
> 
> When I use:
> 
> switch_ivr_play_file(session, NULL, file_path, NULL);
> 
> to play a file such as "misc/8000/sorry.wav" directly, the sound is so choppy.
> 
> Am I doing the background sound playing the right way?
> 
> Han
> --
> 
> 
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> 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-d
> ev
> http://www.freeswitch.org
> 
> !DSPAM:4ea6780b32761207110701!
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> 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-d
> ev
> http://www.freeswitch.org
> 

-- 

 



From peter.olsson at visionutveckling.se  Tue Oct 25 13:50:59 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 25 Oct 2011 11:50:59 +0200
Subject: [Freeswitch-dev] choppy sound with .wav file
	played	inthebackground
In-Reply-To: <ed51f8e938d7265940f67abb91017385@gw.ind.rwth-aachen.de>
References: <549CFEF87AEDE841A38E9D15EAB4C04C59F2AEC8D0@cooper>
	<ed51f8e938d7265940f67abb91017385@gw.ind.rwth-aachen.de>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C59F2AEC950@cooper>

For a bridged call you're not supposed to use switch_ivr_play_file(), since the bridge is sending frames in a separate thread. I think switch_ivr_displace_session() should work for your needs.

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] F?r Han Xin
Skickat: den 25 oktober 2011 11:24
Till: freeswitch-dev at lists.freeswitch.org
?mne: Re: [Freeswitch-dev] choppy sound with .wav file played inthebackground

 hmm, just one call with two sessions, one is the local session, the other is the remote/partner session.

I want to play a indication sound file when I press a button on each side, while the call is ongoing. That's what I mean with
the background. sorry for the vague description.

Regards,
Han
 
----------------original message-----------------
From: "Peter Olsson" 
To: "'freeswitch-dev at lists.freeswitch.org '" freeswitch-dev at lists.freeswitch.org 
Date: Tue, 25 Oct 2011 11:00:53 +0200
-------------------------------------------------
 
 
> How do you mean when you say "in background"? Are ju running this command from 
> another thread then the channel's own thread?
> 
> /Peter
> 
> 
> -----Ursprungligt meddelande-----
> Fr?n: freeswitch-dev-bounces at lists.freeswitch.org 
> [mailto:freeswitch-dev-bounces at lists.freeswitch.org ] F?r Han Xin
> Skickat: den 25 oktober 2011 10:53
> Till: freeswitch-dev at lists.freeswitch.org 
> ?mne: [Freeswitch-dev] choppy sound with .wav file played in the background
> 
> Hi all, I have another question to ask again :P
> 
> When I use:
> 
> switch_ivr_play_file(session, NULL, file_path, NULL);
> 
> to play a file such as "misc/8000/sorry.wav" directly, the sound is so choppy.
> 
> Am I doing the background sound playing the right way?
> 
> Han
> --
> 
> 
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> 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-d
> ev
> http://www.freeswitch.org
> 
> 
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> 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-d
> ev
> http://www.freeswitch.org
> 

-- 

 


_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
http://www.cluecon.com 877-7-4ACLUE

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:4ea67f5932765780018021!


From xin at ind.rwth-aachen.de  Tue Oct 25 18:38:24 2011
From: xin at ind.rwth-aachen.de (Han Xin)
Date: Tue, 25 Oct 2011 16:38:24 +0200
Subject: [Freeswitch-dev] choppy sound with .wav
	fileplayedinthebackground
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C59F2AEC950@cooper>
Message-ID: <590565f7593a9180dbe9caf8bb7a31da@gw.ind.rwth-aachen.de>

 switch_ivr_displace_session() works!

Thanks a lot!
 
----------------original message-----------------
From: "Peter Olsson" 
To: "'freeswitch-dev at lists.freeswitch.org '" freeswitch-dev at lists.freeswitch.org 
Date: Tue, 25 Oct 2011 11:50:59 +0200
-------------------------------------------------
 
 
> For a bridged call you're not supposed to use switch_ivr_play_file(), since the 
> bridge is sending frames in a separate thread. I think 
> switch_ivr_displace_session() should work for your needs.
> 
> /Peter
> 
> 
> -----Ursprungligt meddelande-----
> Fr?n: freeswitch-dev-bounces at lists.freeswitch.org 
> [mailto:freeswitch-dev-bounces at lists.freeswitch.org ] F?r Han Xin
> Skickat: den 25 oktober 2011 11:24
> Till: freeswitch-dev at lists.freeswitch.org 
> ?mne: Re: [Freeswitch-dev] choppy sound with .wav file played 
> inthebackground
> 
> hmm, just one call with two sessions, one is the local session, the other is the 
> remote/partner session.
> 
> I want to play a indication sound file when I press a button on each side, while the 
> call is ongoing. That's what I mean with
> the background. sorry for the vague description.
> 
> Regards,
> Han
> 
> ----------------original message-----------------
> From: "Peter Olsson" 
> To: "'freeswitch-dev at lists.freeswitch.org '" 
> freeswitch-dev at lists.freeswitch.org 
> Date: Tue, 25 Oct 2011 11:00:53 +0200
> -------------------------------------------------
> 
> 
>> How do you mean when you say "in background"? Are ju running this command from 
>> another thread then the channel's own thread?
>> 
>> /Peter
>> 
>> 
>> -----Ursprungligt meddelande-----
>> Fr?n: freeswitch-dev-bounces at lists.freeswitch.org 
>> [mailto:freeswitch-dev-bounces at lists.freeswitch.org ] F?r Han Xin
>> Skickat: den 25 oktober 2011 10:53
>> Till: freeswitch-dev at lists.freeswitch.org 
>> ?mne: [Freeswitch-dev] choppy sound with .wav file played in the 
>> background
>> 
>> Hi all, I have another question to ask again :P
>> 
>> When I use:
>> 
>> switch_ivr_play_file(session, NULL, file_path, NULL);
>> 
>> to play a file such as "misc/8000/sorry.wav" directly, the sound is so 
>> choppy.
>> 
>> Am I doing the background sound playing the right way?
>> 
>> Han
>> --
>> 
>> 
>> 
>> 
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>> 
>> 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
>> -d
>> ev
>> http://www.freeswitch.org
>> 
>> 
>> 
>> 
>> _______________________________________________
>> Join us at ClueCon 2011, Aug 9-11, Chicago
>> http://www.cluecon.com 877-7-4ACLUE
>> 
>> 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
>> -d
>> ev
>> http://www.freeswitch.org
>> 
> 
> -- 
> 
> 
> 
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> 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-d
> ev
> http://www.freeswitch.org
> 
> !DSPAM:4ea67f5932765780018021!
> 
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
> 
> 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-d
> ev
> http://www.freeswitch.org
> 

-- 

 



From anthony.minessale at gmail.com  Tue Oct 25 18:50:44 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 25 Oct 2011 09:50:44 -0500
Subject: [Freeswitch-dev] switch_core_sqldb.c
In-Reply-To: <FEBDE042-CD20-4BD3-93D9-5827EFA49F83@gmail.com>
References: <BANLkTim5jOf70cX52n4VJeqNrDWoxf2iHg@mail.gmail.com>
	<FEBDE042-CD20-4BD3-93D9-5827EFA49F83@gmail.com>
Message-ID: <CAKbxfG-MdU_43Rgn7YeFmHNCtahZVvn8nhQJMGSaC745FiqeBw@mail.gmail.com>

I think that was already fixed.
It always pays to update to GIT HEAD when in doubt.


On Sun, Jul 17, 2011 at 2:39 PM, Robert Huddleston <rhuddleston at gmail.com>wrote:

> Wondering if I should log this to Jira?
>
> Looks like a recent code addition - so not sure protocol here - also never
> having logged a Jira before.
>
> In switch_core_sqldb.c there are some PRAGMA calls.
>
> I'm finding that in my MYSQL unixODBC environment - these PRAGMA calls are
> causing syntax errors.
>
> I'm not familiar with the syntax - I've seen equivalents on MS-SQL and
> Oracle etc - but never seen such on MYSQL.
>
>
> http://fisheye.freeswitch.org/changelog/freeswitch.git/?showid=11690aff4c4b819e981c4763abff37c33700f5ab&view=fe
>
> Any ideas?
>
>        switch_cache_db_execute_sql(sql_manager.event_db, "PRAGMA
> synchronous=OFF;", NULL);
>        switch_cache_db_execute_sql(sql_manager.event_db, "PRAGMA
> count_changes=OFF;", NULL);
>        switch_cache_db_execute_sql(sql_manager.event_db, "PRAGMA
> temp_store=MEMORY;", NULL);
>        switch_cache_db_execute_sql(sql_manager.event_db, "PRAGMA
> journal_mode=OFF;", NULL);
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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
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From callum.guy at x-on.co.uk  Tue Oct 25 13:52:09 2011
From: callum.guy at x-on.co.uk (Callum Guy)
Date: Tue, 25 Oct 2011 10:52:09 +0100
Subject: [Freeswitch-dev] choppy sound with .wav file played
	inthebackground
In-Reply-To: <ed51f8e938d7265940f67abb91017385@gw.ind.rwth-aachen.de>
References: <549CFEF87AEDE841A38E9D15EAB4C04C59F2AEC8D0@cooper>
	<ed51f8e938d7265940f67abb91017385@gw.ind.rwth-aachen.de>
Message-ID: <CAFjCFz=9jv3wOTX_yP3CWavH5rpu6_c-CpffGxMimPXr7zeQrQ@mail.gmail.com>

Han,

I think i may have come across this sort of issue before with my first use
of the FreeSWITCH system.

Have you seen anything such as this in the logs/fs_cli output during the
call?

2011-09-28 15:37:12.047658 [DEBUG] switch_ivr.c:577 sofia/internal/1000
Command Execute
playback(/usr/local/freeswitch/sounds/en/uk/cg_test/cg_out_test_welcome.wav)
EXECUTE sofia/internal/1000
playback(/usr/local/freeswitch/sounds/en/uk/cg_test/cg_out_test_welcome.wav)
2011-09-28 15:37:12.047658 [DEBUG] switch_ivr_play_say.c:1302 Codec
Activated L16 at 8000hz 1 channels 30ms
2011-09-28 15:37:12.107661 [DEBUG] switch_rtp.c:3181 Correct ip/port
confirmed.
2011-09-28 15:37:12.317674 [WARNING] mod_sofia.c:1115 Asynchronous PTIME not
supported, changing our end from 30 to 20
2011-09-28 15:37:12.317674 [DEBUG] sofia_glue.c:2775 Changing Codec from
PCMA at 30ms@8000hz to PCMA at 20ms@8000hz
2011-09-28 15:37:12.367677 [DEBUG] switch_rtp.c:1527 RE-Starting timer
[soft] 160 bytes per 20ms
2011-09-28 15:37:12.367677 [DEBUG] sofia_glue.c:2857 Set Codec
sofia/internal/1000 PCMA/8000 20 ms 160 samples 64000 bits

I looked into this at the time and my understanding was that I need to try
and add PCMA at 20i in /usr/local/src/freeswitch/conf/vars.xml and re-compile
however i have not revisited the issue yet as i only found this to be an
issue with outbound only calls from FS, something i was doing just as a
demonstration.

I found it can be resolved by updating my handset to set RTP Packet Size to
0.020 (its default was 0.030). I doubt this is of much help but perhaps its
worth checking? This is my first contribution to the mailing list and i'm
not a greatly experienced used but i did want to raise this as a possibility
- apologies if this hasn't been of any use!

Thanks,

Callum Guy


On 25 October 2011 10:24, Han Xin <xin at ind.rwth-aachen.de> wrote:

>  hmm, just one call with two sessions, one is the local session, the other
> is the remote/partner session.
>
> I want to play a indication sound file when I press a button on each side,
> while the call is ongoing. That's what I mean with
> the background. sorry for the vague description.
>
> Regards,
> Han
>
> ----------------original message-----------------
> From: "Peter Olsson"
> To: "'freeswitch-dev at lists.freeswitch.org '"
> freeswitch-dev at lists.freeswitch.org
> Date: Tue, 25 Oct 2011 11:00:53 +0200
> -------------------------------------------------
>
>
> > How do you mean when you say "in background"? Are ju running this command
> from
> > another thread then the channel's own thread?
> >
> > /Peter
> >
> >
> > -----Ursprungligt meddelande-----
> > Fr?n: freeswitch-dev-bounces at lists.freeswitch.org
> > [mailto:freeswitch-dev-bounces at lists.freeswitch.org ] F?r Han Xin
> > Skickat: den 25 oktober 2011 10:53
> > Till: freeswitch-dev at lists.freeswitch.org
> > ?mne: [Freeswitch-dev] choppy sound with .wav file played in the
> background
> >
> > Hi all, I have another question to ask again :P
> >
> > When I use:
> >
> > switch_ivr_play_file(session, NULL, file_path, NULL);
> >
> > to play a file such as "misc/8000/sorry.wav" directly, the sound is so
> choppy.
> >
> > Am I doing the background sound playing the right way?
> >
> > Han
> > --
> >
> >
> >
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > 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-d
> > ev
> > http://www.freeswitch.org
> >
> > !DSPAM:4ea6780b32761207110701!
> >
> >
> > _______________________________________________
> > Join us at ClueCon 2011, Aug 9-11, Chicago
> > http://www.cluecon.com 877-7-4ACLUE
> >
> > 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-d
> > ev
> > http://www.freeswitch.org
> >
>
> --
>
>
>
>
> _______________________________________________
> Join us at ClueCon 2011, Aug 9-11, Chicago
> http://www.cluecon.com 877-7-4ACLUE
>
> 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 xin at ind.rwth-aachen.de  Wed Oct 26 11:34:12 2011
From: xin at ind.rwth-aachen.de (Han Xin)
Date: Wed, 26 Oct 2011 09:34:12 +0200
Subject: [Freeswitch-dev] choppy sound with .wav file played in the
 background
In-Reply-To: <CAFjCFz=9jv3wOTX_yP3CWavH5rpu6_c-CpffGxMimPXr7zeQrQ@mail.gmail.com>
Message-ID: <cc7b9ddb4c1746f2c753b6078078dd9f@gw.ind.rwth-aachen.de>

Hi Callum,

Thanks for the detailed reply first! My problem was simply solved by using another
function "switch_ivr_displace_session()" to play the indication sound file. 

Anyway it is very nice of you to provide this solution, other developers may get 
benefit from your contribution. :)

Regards,
Han
 
----------------urspr?ngliche Nachricht-----------------
Von: "Callum Guy" callum.guy at x-on.co.uk 
An: freeswitch-dev at lists.freeswitch.org 
Datum: Tue, 25 Oct 2011 10:52:09 +0100
-------------------------------------------------
 
 
> Han,
> 
> I think i may have come across this sort of issue before with my first use
> of the FreeSWITCH system.
> 
> Have you seen anything such as this in the logs/fs_cli output during the
> call?
> 
> 2011-09-28 15:37:12.047658 [DEBUG] switch_ivr.c:577 sofia/internal/1000
> Command Execute
> 
> playback(/usr/local/freeswitch/sounds/en/uk/cg_test/cg_out_test_welc
> ome.wav)
> EXECUTE sofia/internal/1000
> 
> playback(/usr/local/freeswitch/sounds/en/uk/cg_test/cg_out_test_welc
> ome.wav)
> 2011-09-28 15:37:12.047658 [DEBUG] switch_ivr_play_say.c:1302 Codec
> Activated L16 at 8000hz 1 channels 30ms
> 2011-09-28 15:37:12.107661 [DEBUG] switch_rtp.c:3181 Correct ip/port
> confirmed.
> 2011-09-28 15:37:12.317674 [WARNING] mod_sofia.c:1115 Asynchronous PTIME 
> not
> supported, changing our end from 30 to 20
> 2011-09-28 15:37:12.317674 [DEBUG] sofia_glue.c:2775 Changing Codec from
> PCMA at 30ms @8000hz to PCMA at 20ms @8000hz
> 2011-09-28 15:37:12.367677 [DEBUG] switch_rtp.c:1527 RE-Starting timer
> [soft] 160 bytes per 20ms
> 2011-09-28 15:37:12.367677 [DEBUG] sofia_glue.c:2857 Set Codec
> sofia/internal/1000 PCMA/8000 20 ms 160 samples 64000 bits
> 
> I looked into this at the time and my understanding was that I need to try
> and add PCMA at 20i in /usr/local/src/freeswitch/conf/vars.xml and 
> re-compile
> however i have not revisited the issue yet as i only found this to be an
> issue with outbound only calls from FS, something i was doing just as a
> demonstration.
> 
> I found it can be resolved by updating my handset to set RTP Packet Size to
> 0.020 (its default was 0.030). I doubt this is of much help but perhaps its
> worth checking? This is my first contribution to the mailing list and i'm
> not a greatly experienced used but i did want to raise this as a possibility
> - apologies if this hasn't been of any use!
> 
> Thanks,
> 
> Callum Guy
> 
> 
> On 25 October 2011 10:24, Han Xin xin at ind.rwth-aachen.de wrote:
> 
>> hmm, just one call with two sessions, one is the local session, the other
>> is the remote/partner session.
>>
>> I want to play a indication sound file when I press a button on each side,
>> while the call is ongoing. That's what I mean with
>> the background. sorry for the vague description.
>>
>> Regards,
>> Han
>>
>> ----------------original message-----------------
>> From: "Peter Olsson"
>> To: "'freeswitch-dev at lists.freeswitch.org '"
>> freeswitch-dev at lists.freeswitch.org 
>> Date: Tue, 25 Oct 2011 11:00:53 +0200
>> -------------------------------------------------
>>
>>
>> > How do you mean when you say "in background"? Are ju running this command
>> from
>> > another thread then the channel's own thread?
>> >
>> > /Peter
>> >
>> >
>> > -----Ursprungligt meddelande-----
>> > Fr?n: freeswitch-dev-bounces at lists.freeswitch.org 
>> > [mailto:freeswitch-dev-bounces at lists.freeswitch.org ] F?r Han Xin
>> > Skickat: den 25 oktober 2011 10:53
>> > Till: freeswitch-dev at lists.freeswitch.org 
>> > ?mne: [Freeswitch-dev] choppy sound with .wav file played in the
>> background
>> >
>> > Hi all, I have another question to ask again :P
>> >
>> > When I use:
>> >
>> > switch_ivr_play_file(session, NULL, file_path, NULL);
>> >
>> > to play a file such as "misc/8000/sorry.wav" directly, the sound is so
>> choppy.
>> >
>> > Am I doing the background sound playing the right way?
>> >
>> > Han
>> > --
>> >
>> >
>> >
>> >
>> > _______________________________________________
>> > Join us at ClueCon 2011, Aug 9-11, Chicago
>> > http://www.cluecon.com 877-7-4ACLUE
>> >
>> > FreeSWITCH-dev mailing list
>> > FreeSWITCH-dev at lists.freeswitch.org 
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>> >
>> > 
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From victor.velo at videotron.ca  Mon Oct 31 17:43:02 2011
From: victor.velo at videotron.ca (victor.velo at videotron.ca)
Date: Mon, 31 Oct 2011 10:43:02 -0400
Subject: [Freeswitch-dev] joining LegA and LegB and to have the UUID_LegA
 and UUID_LegB accessible
Message-ID: <75208aef16a616.4eae7bb6@videotron.ca>

Hello,

I tried to join two legs (A,B) using the "api uuid_bridge " + legA_uuid + " " + legB_uuid in ESL program. It seems to work, but I could not have access to legA nor to legB anymore after the bridge.

Q: Is there a way to join the two legs that allows me to have access to the two legs after the bridge?

Thanks in advance,

Victor

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From beffa at ieee.org  Mon Oct 31 22:46:38 2011
From: beffa at ieee.org (Federico Beffa)
Date: Mon, 31 Oct 2011 20:46:38 +0100
Subject: [Freeswitch-dev] mod_dingaling timing out
Message-ID: <CAKrPhPNZbBbp2PQfLtvyRZhw=fmsb57EUmFBXq=GBJT1xhYUmw@mail.gmail.com>

Dear Developers,

I'm using a fairly recent version of FS (pulled 20 Oct.) and I'm
experiencing a timeout problem with mod_dingaling:

With FS running on a fairly fast server (Intel Xeon CPU E31220) with
Ubuntu 10.04 server, at an interval of approximately 3h (varying
between 2h and 5h) I see mod_dingaling disconnecting (for some
minutes) and reconnecting to google talk (client mode). To find out
the root cause I've added some debugging lines in the code and have
found that the reported reason is that the function "SSL_read" in the
function "iks_recv" of the FS customized iksemel library returns
SSL_ERROR_SYSCALL. Looking at errno and related message I get: errno =
110 (Connection timed out).

I'm behind NAT, but due to the recently introduced "keep alive"
feature of mod_dingaling that should not be the cause of the problem.

The problem should also not be due to network problems as:

* during a disconnect, from the same machine I could ping
  talk.google.com and "openssl s_client -connect talk.google.com:5222
  -starttls xmpp -debug" replies correctly.

* simultaneously running FS on a different machine as a different user
  shows the same problem, but at different points in time and
  sometimes (?) longer intervals.

Reading the man page for SSL_read I learned that this function can
also write to the socket (as SSL_write can read). I'm wondering if the
use of poll() (in iks_recv) with only read flags in conjunction with
SSL_read could somehow be responsible for this.

I would appreciate any suggestion on how to investigate deeper the
problem.

Thanks,
Fede