From fdelawarde at wirelessmundi.com  Wed Aug  4 08:13:02 2010
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Wed, 04 Aug 2010 17:13:02 +0200
Subject: [Freeswitch-dev] cluecon progress
Message-ID: <1280934782.13885.224.camel@luna.tc.commsmundi.com>

How is it going up there?

Fran?ois.



From jmesquita at freeswitch.org  Wed Aug  4 08:37:22 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 4 Aug 2010 12:37:22 -0300
Subject: [Freeswitch-dev] cluecon progress
In-Reply-To: <1280934782.13885.224.camel@luna.tc.commsmundi.com>
References: <1280934782.13885.224.camel@luna.tc.commsmundi.com>
Message-ID: <AANLkTinH+7O0b9dvkk1Dtvj6a2eG-6PpHRYNc_sMZJt0@mail.gmail.com>

Please, fill us in!! We are all anxious to know.

Regards,
Jo?o Mesquita
FreeSWITCH? Solutions


On Wed, Aug 4, 2010 at 12:13 PM, Fran?ois Delawarde <
fdelawarde at wirelessmundi.com> wrote:

> How is it going up there?
>
> Fran?ois.
>
>
> _______________________________________________
> 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 gmaruzz at celliax.org  Wed Aug  4 08:41:46 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 4 Aug 2010 17:41:46 +0200
Subject: [Freeswitch-dev] cluecon progress
In-Reply-To: <1280934782.13885.224.camel@luna.tc.commsmundi.com>
References: <1280934782.13885.224.camel@luna.tc.commsmundi.com>
Message-ID: <AANLkTik2o2xN4xPaO2VTMXzd613Crvk1NeW_5DzsztoV@mail.gmail.com>

On Wed, Aug 4, 2010 at 5:13 PM, Fran?ois Delawarde
<fdelawarde at wirelessmundi.com> wrote:
> How is it going up there?

Yep, yep, let us know!

-giovanni

-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From msc at freeswitch.org  Wed Aug  4 10:02:59 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 4 Aug 2010 12:02:59 -0500
Subject: [Freeswitch-dev] cluecon progress
In-Reply-To: <AANLkTinH+7O0b9dvkk1Dtvj6a2eG-6PpHRYNc_sMZJt0@mail.gmail.com>
References: <1280934782.13885.224.camel@luna.tc.commsmundi.com>
	<AANLkTinH+7O0b9dvkk1Dtvj6a2eG-6PpHRYNc_sMZJt0@mail.gmail.com>
Message-ID: <AANLkTikk4sEaznovpjhduuHv2XVqs-P_XJ62y78ECY70@mail.gmail.com>

AWESOME! :P

2010/8/4 Jo?o Mesquita <jmesquita at freeswitch.org>

> Please, fill us in!! We are all anxious to know.
>
> Regards,
> Jo?o Mesquita
> FreeSWITCH? Solutions
>
>
>
> On Wed, Aug 4, 2010 at 12:13 PM, Fran?ois Delawarde <
> fdelawarde at wirelessmundi.com> wrote:
>
>> How is it going up there?
>>
>> Fran?ois.
>>
>>
>> _______________________________________________
>> 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
>>
>
>
> _______________________________________________
> 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 jmesquita at freeswitch.org  Wed Aug  4 10:18:55 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 4 Aug 2010 14:18:55 -0300
Subject: [Freeswitch-dev] cluecon progress
In-Reply-To: <AANLkTikk4sEaznovpjhduuHv2XVqs-P_XJ62y78ECY70@mail.gmail.com>
References: <1280934782.13885.224.camel@luna.tc.commsmundi.com>
	<AANLkTinH+7O0b9dvkk1Dtvj6a2eG-6PpHRYNc_sMZJt0@mail.gmail.com>
	<AANLkTikk4sEaznovpjhduuHv2XVqs-P_XJ62y78ECY70@mail.gmail.com>
Message-ID: <AANLkTinnKNNGFR0s3drdzFgf=GNS3eb0rAmR8qDYuMk1@mail.gmail.com>

You mean....
Jo?o Mesquita
FreeSWITCH? Solutions


On Wed, Aug 4, 2010 at 2:02 PM, Michael Collins <msc at freeswitch.org> wrote:

> AWESOME! :P
>
> 2010/8/4 Jo?o Mesquita <jmesquita at freeswitch.org>
>
> Please, fill us in!! We are all anxious to know.
>>
>> Regards,
>> Jo?o Mesquita
>> FreeSWITCH? Solutions
>>
>>
>>
>> On Wed, Aug 4, 2010 at 12:13 PM, Fran?ois Delawarde <
>> fdelawarde at wirelessmundi.com> wrote:
>>
>>> How is it going up there?
>>>
>>> Fran?ois.
>>>
>>>
>>> _______________________________________________
>>> 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
>>>
>>
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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 tayeb.meftah at gmail.com  Thu Aug  5 11:53:10 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Thu, 05 Aug 2010 20:53:10 +0200
Subject: [Freeswitch-dev] cluecon progress
In-Reply-To: <AANLkTikk4sEaznovpjhduuHv2XVqs-P_XJ62y78ECY70@mail.gmail.com>
References: <1280934782.13885.224.camel@luna.tc.commsmundi.com>	<AANLkTinH+7O0b9dvkk1Dtvj6a2eG-6PpHRYNc_sMZJt0@mail.gmail.com>
	<AANLkTikk4sEaznovpjhduuHv2XVqs-P_XJ62y78ECY70@mail.gmail.com>
Message-ID: <4C5B0896.2080601@gmail.com>

:P is not enoug!
report to us what's new!
Le 04/08/2010 19:02, Michael Collins a ?crit :
> AWESOME! :P
>
> 2010/8/4 Jo?o Mesquita <jmesquita at freeswitch.org 
> <mailto:jmesquita at freeswitch.org>>
>
>     Please, fill us in!! We are all anxious to know.
>
>     Regards,
>     Jo?o Mesquita
>     FreeSWITCH? Solutions
>
>
>
>     On Wed, Aug 4, 2010 at 12:13 PM, Fran?ois Delawarde
>     <fdelawarde at wirelessmundi.com
>     <mailto:fdelawarde at wirelessmundi.com>> wrote:
>
>         How is it going up there?
>
>         Fran?ois.
>
>
>         _______________________________________________
>         FreeSWITCH-dev mailing list
>         FreeSWITCH-dev at lists.freeswitch.org
>         <mailto:FreeSWITCH-dev at lists.freeswitch.org>
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>         http://www.freeswitch.org
>
>
>
>     _______________________________________________
>     FreeSWITCH-dev mailing list
>     FreeSWITCH-dev at lists.freeswitch.org
>     <mailto:FreeSWITCH-dev at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> 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
>    


-- 
Meftah Tayeb
alg?rie t?l?com SPA
phone: +21321761805
phone (INUM): +883510001289101
mobile : +213660347746
mobile (INUM: +883510001289110
http://www.algerietelecom.dz

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From gmaruzz at celliax.org  Wed Aug  4 13:17:47 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 4 Aug 2010 22:17:47 +0200
Subject: [Freeswitch-dev] cluecon progress
In-Reply-To: <AANLkTikk4sEaznovpjhduuHv2XVqs-P_XJ62y78ECY70@mail.gmail.com>
References: <1280934782.13885.224.camel@luna.tc.commsmundi.com> 
	<AANLkTinH+7O0b9dvkk1Dtvj6a2eG-6PpHRYNc_sMZJt0@mail.gmail.com> 
	<AANLkTikk4sEaznovpjhduuHv2XVqs-P_XJ62y78ECY70@mail.gmail.com>
Message-ID: <AANLkTimRubCCdeVYk7zmF0bJth+ydiR9x6zQo1RgPw7y@mail.gmail.com>

On Wed, Aug 4, 2010 at 7:02 PM, Michael Collins <msc at freeswitch.org> wrote:
> AWESOME! :P

Yeah...


>
> 2010/8/4 Jo?o Mesquita <jmesquita at freeswitch.org>
>>
>> Please, fill us in!! We are all anxious to know.
>> Regards,
>> Jo?o Mesquita
>> FreeSWITCH? Solutions
>>
>>
>> On Wed, Aug 4, 2010 at 12:13 PM, Fran?ois Delawarde
>> <fdelawarde at wirelessmundi.com> wrote:
>>>
>>> How is it going up there?
>>>
>>> Fran?ois.
>>>
>>>
>>> _______________________________________________
>>> 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
>>
>>
>> _______________________________________________
>> 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
>>
>
>
> _______________________________________________
> 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
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From brian at freeswitch.org  Wed Aug  4 13:22:09 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 4 Aug 2010 15:22:09 -0500
Subject: [Freeswitch-dev] cluecon progress
In-Reply-To: <AANLkTimRubCCdeVYk7zmF0bJth+ydiR9x6zQo1RgPw7y@mail.gmail.com>
References: <1280934782.13885.224.camel@luna.tc.commsmundi.com>
	<AANLkTinH+7O0b9dvkk1Dtvj6a2eG-6PpHRYNc_sMZJt0@mail.gmail.com>
	<AANLkTikk4sEaznovpjhduuHv2XVqs-P_XJ62y78ECY70@mail.gmail.com>
	<AANLkTimRubCCdeVYk7zmF0bJth+ydiR9x6zQo1RgPw7y@mail.gmail.com>
Message-ID: <9B648900-8D3F-4E1E-9EEE-A07AE4D4FB3E@freeswitch.org>

Should have been here!  ;)

/b

On Aug 4, 2010, at 3:17 PM, Giovanni Maruzzelli wrote:

> On Wed, Aug 4, 2010 at 7:02 PM, Michael Collins <msc at freeswitch.org> wrote:
>> AWESOME! :P
> 
> Yeah...
> 

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From herrold at owlriver.com  Wed Aug  4 14:35:35 2010
From: herrold at owlriver.com (R P Herrold)
Date: Wed, 4 Aug 2010 17:35:35 -0400 (EDT)
Subject: [Freeswitch-dev] cluecon progress
In-Reply-To: <9B648900-8D3F-4E1E-9EEE-A07AE4D4FB3E@freeswitch.org>
References: <1280934782.13885.224.camel@luna.tc.commsmundi.com>
	<AANLkTinH+7O0b9dvkk1Dtvj6a2eG-6PpHRYNc_sMZJt0@mail.gmail.com>
	<AANLkTikk4sEaznovpjhduuHv2XVqs-P_XJ62y78ECY70@mail.gmail.com>
	<AANLkTimRubCCdeVYk7zmF0bJth+ydiR9x6zQo1RgPw7y@mail.gmail.com>
	<9B648900-8D3F-4E1E-9EEE-A07AE4D4FB3E@freeswitch.org>
Message-ID: <alpine.LRH.1.00.1008041734120.4845@oebafba.bjyevire.pbz>

On Wed, 4 Aug 2010, Brian West wrote:

> Should have been here!  ;)
>
> /b

mean Brian

(you not here will have to excuse him -- he seems to be 
starving, and is wasting away)

I've been trying to post every hour or so on my twitter feed:
 	http://twitter.com/herrold
with the #cluecon tag

-- Russ herrold


From jaybinks at gmail.com  Wed Aug  4 13:22:18 2010
From: jaybinks at gmail.com (jay binks)
Date: Thu, 5 Aug 2010 06:22:18 +1000
Subject: [Freeswitch-dev] cluecon progress
In-Reply-To: <4C5B0896.2080601@gmail.com>
References: <1280934782.13885.224.camel@luna.tc.commsmundi.com> 
	<AANLkTinH+7O0b9dvkk1Dtvj6a2eG-6PpHRYNc_sMZJt0@mail.gmail.com> 
	<AANLkTikk4sEaznovpjhduuHv2XVqs-P_XJ62y78ECY70@mail.gmail.com> 
	<4C5B0896.2080601@gmail.com>
Message-ID: <AANLkTinvgQSN=i65YuJkq5Z7xFCCU0QDDZ1yxqoef=2c@mail.gmail.com>

HA Clustering with live call recovery is new :)

On Fri, Aug 6, 2010 at 4:53 AM, Meftah Tayeb <tayeb.meftah at gmail.com> wrote:

>  :P is not enoug!
> report to us what's new!
>
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From brian at freeswitch.org  Thu Aug  5 06:40:13 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 5 Aug 2010 08:40:13 -0500
Subject: [Freeswitch-dev] cluecon progress
In-Reply-To: <alpine.LRH.1.00.1008041734120.4845@oebafba.bjyevire.pbz>
References: <1280934782.13885.224.camel@luna.tc.commsmundi.com>
	<AANLkTinH+7O0b9dvkk1Dtvj6a2eG-6PpHRYNc_sMZJt0@mail.gmail.com>
	<AANLkTikk4sEaznovpjhduuHv2XVqs-P_XJ62y78ECY70@mail.gmail.com>
	<AANLkTimRubCCdeVYk7zmF0bJth+ydiR9x6zQo1RgPw7y@mail.gmail.com>
	<9B648900-8D3F-4E1E-9EEE-A07AE4D4FB3E@freeswitch.org>
	<alpine.LRH.1.00.1008041734120.4845@oebafba.bjyevire.pbz>
Message-ID: <A0703C36-7B38-4C9C-81E3-804A757AD796@freeswitch.org>

This year the WiFi didn't fall apart and everyone seems to be very happy with the Trump.

/b

On Aug 4, 2010, at 4:35 PM, R P Herrold wrote:

> mean Brian
> 
> (you not here will have to excuse him -- he seems to be 
> starving, and is wasting away)
> 
> I've been trying to post every hour or so on my twitter feed:
> 	http://twitter.com/herrold
> with the #cluecon tag
> 
> -- Russ herrold

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From jaybinks at gmail.com  Thu Aug  5 07:15:51 2010
From: jaybinks at gmail.com (jay binks)
Date: Fri, 6 Aug 2010 00:15:51 +1000
Subject: [Freeswitch-dev] cluecon progress
In-Reply-To: <A0703C36-7B38-4C9C-81E3-804A757AD796@freeswitch.org>
References: <1280934782.13885.224.camel@luna.tc.commsmundi.com> 
	<AANLkTinH+7O0b9dvkk1Dtvj6a2eG-6PpHRYNc_sMZJt0@mail.gmail.com> 
	<AANLkTikk4sEaznovpjhduuHv2XVqs-P_XJ62y78ECY70@mail.gmail.com> 
	<AANLkTimRubCCdeVYk7zmF0bJth+ydiR9x6zQo1RgPw7y@mail.gmail.com> 
	<9B648900-8D3F-4E1E-9EEE-A07AE4D4FB3E@freeswitch.org>
	<alpine.LRH.1.00.1008041734120.4845@oebafba.bjyevire.pbz> 
	<A0703C36-7B38-4C9C-81E3-804A757AD796@freeswitch.org>
Message-ID: <AANLkTinaRc0OUsbtSiv3PD8YtX0kf1CuQa3zOutBecoK@mail.gmail.com>

this year... the WIFI is exceptionally stable...  VERY Good.

and Trump is awesome, cant top it...
and the conference rate @ Trump is amazing...   $225 per night is VERY
reasonable
I was expecting double that for trump.

I cant imagine why you'd invest time in finding an alternate location :P

this years ClueCon has been near perfect


J



On Thu, Aug 5, 2010 at 11:40 PM, Brian West <brian at freeswitch.org> wrote:

> This year the WiFi didn't fall apart and everyone seems to be very happy
> with the Trump.
>
> /b
>
> On Aug 4, 2010, at 4:35 PM, R P Herrold wrote:
>
> mean Brian
>
> (you not here will have to excuse him -- he seems to be
> starving, and is wasting away)
>
> I've been trying to post every hour or so on my twitter feed:
> http://twitter.com/herrold
> with the #cluecon tag
>
> -- Russ herrold
>
>
>
> _______________________________________________
> 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
>
>


-- 
Sincerely

Jay
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From brian at freeswitch.org  Thu Aug  5 07:33:28 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 5 Aug 2010 09:33:28 -0500
Subject: [Freeswitch-dev] cluecon progress
In-Reply-To: <AANLkTinaRc0OUsbtSiv3PD8YtX0kf1CuQa3zOutBecoK@mail.gmail.com>
References: <1280934782.13885.224.camel@luna.tc.commsmundi.com>
	<AANLkTinH+7O0b9dvkk1Dtvj6a2eG-6PpHRYNc_sMZJt0@mail.gmail.com>
	<AANLkTikk4sEaznovpjhduuHv2XVqs-P_XJ62y78ECY70@mail.gmail.com>
	<AANLkTimRubCCdeVYk7zmF0bJth+ydiR9x6zQo1RgPw7y@mail.gmail.com>
	<9B648900-8D3F-4E1E-9EEE-A07AE4D4FB3E@freeswitch.org>
	<alpine.LRH.1.00.1008041734120.4845@oebafba.bjyevire.pbz>
	<A0703C36-7B38-4C9C-81E3-804A757AD796@freeswitch.org>
	<AANLkTinaRc0OUsbtSiv3PD8YtX0kf1CuQa3zOutBecoK@mail.gmail.com>
Message-ID: <7C0FC64F-A65B-4882-9E07-11327107C82F@freeswitch.org>

It usually is double that.

/b

On Aug 5, 2010, at 9:15 AM, jay binks wrote:

> I was expecting double that for trump.



From msc at freeswitch.org  Wed Aug 11 09:39:19 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 11 Aug 2010 09:39:19 -0700
Subject: [Freeswitch-dev] FreeSWITCH Conference Call Starting Shortly!
Message-ID: <AANLkTinYeaKPAkX9MzZd585dmvsT4Y=d8JTgGOb6pE9E@mail.gmail.com>

Hello all,

Please be sure to join the conf call. The agenda is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2010_08_11

It is light today but that will give us time to recap some of the fun stuff
that happened at ClueCon. :P

Thanks!

-MC
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From freeswitch-list at puzzled.xs4all.nl  Wed Aug 11 12:15:55 2010
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Wed, 11 Aug 2010 21:15:55 +0200
Subject: [Freeswitch-dev] mod_osp: undefined symbol
	OSPPTransactionSetSrcNetworkId**
Message-ID: <4C62F6EB.7090306@puzzled.xs4all.nl>

Hi,

Platform:   CentOS 5.5 x86_64 (updated)
FreeSWITCH: git-07b8176 2010-08-10 18-51-06 -0400
OSPToolkit: 3.6.1

After installing OSPToolkit I enabled mod_osp and built and installed 
FreeSWITCH. When I issue load mod_osp I see this error:

2010-08-11 21:00:29.284157 [CRIT] switch_loadable_module.c:926 Error 
Loading module /opt/freeswitch/mod/mod_osp.so
**/opt/freeswitch/mod/mod_osp.so: undefined symbol: 
OSPPTransactionSetSrcNetworkId**

I'm out of my league here but I tried to get as much info as possible.

/usr/lib64/osp/libosptk.a has the following symbols:

$ nm /usr/lib64/libosptk.a
[snip]
osptransapi.o:
00000000000006ed T OSPPTransactionSetSrcNetworkId
[snip]

Checking mod_osp I see the following:

$ nm /opt/freeswitch/mod/mod_osp.so
0000000000209570 d B64CACert
0000000000209568 d B64LCert
0000000000209560 d B64PKey
                  U OSPPBase64Decode
                  U OSPPCleanup
                  U OSPPInit
                  U OSPPProviderDelete
                  U OSPPProviderNew
                  U OSPPTransactionBuildUsageFromScratch
                  U OSPPTransactionDelete
                  U OSPPTransactionGetContext
                  U OSPPTransactionGetDestProtocol
                  U OSPPTransactionGetDestinationNetworkId
                  U OSPPTransactionGetFirstDestination
                  U OSPPTransactionGetNextDestination
                  U OSPPTransactionGetNumberPortabilityParameters
                  U OSPPTransactionGetOperatorName
                  U OSPPTransactionIsDestOSPEnabled
                  U OSPPTransactionNew
                  U OSPPTransactionRecordFailure
                  U OSPPTransactionReportUsage
                  U OSPPTransactionRequestAuthorisation
                  U OSPPTransactionSetCustomInfo
                  U OSPPTransactionSetDestNetworkId
                  U OSPPTransactionSetDestinationCount
                  U OSPPTransactionSetDiversion
                  U OSPPTransactionSetForwardCodec
                  U OSPPTransactionSetNetworkIds
                  U OSPPTransactionSetNumberPortability
                  U OSPPTransactionSetOctets
                  U OSPPTransactionSetPackets
                  U OSPPTransactionSetReverseCodec
                  U OSPPTransactionSetServiceType
                  U OSPPTransactionSetSrcNetworkId  <--
[snip]

According to the nm manpage, the "U" means "undefined" so mod_osp does 
not seem to pick up libosptk.a (during linking?).

Does anyone perhaps have a suggestion for a fix so I can file a bug with 
a fix?

Thanks!
Patrick


From sergio.alecha at gmail.com  Fri Aug  6 12:03:50 2010
From: sergio.alecha at gmail.com (Sergio Alecha)
Date: Fri, 6 Aug 2010 16:03:50 -0300
Subject: [Freeswitch-dev] Skinny bug
Message-ID: <AANLkTinpi8rR5K-ohhfDxQoJ7LAJyt+NHpUtHHJXuH6g@mail.gmail.com>

I can not make the call transfer module mod_skinny. I am using a cisco 7910
phone. and FreeSwitch version 1.0.6.
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From aroumie at yahoo.com  Tue Aug 10 19:30:33 2010
From: aroumie at yahoo.com (Ali Roumie)
Date: Tue, 10 Aug 2010 19:30:33 -0700 (PDT)
Subject: [Freeswitch-dev] Socket - bridge answered event
Message-ID: <250201.55875.qm@web120608.mail.ne1.yahoo.com>

Hello All, 
This is my first post to this list and many thanks to all contributors to this 
state of art project. I'm using outbound socket and everything is going 
wonderful with me except one thing so far.
My logic is simple, collect a PIN from Leg A and once the PIN is authorized, I 
bridge the call with a SIP provider. Everything is great, call got bridged 
successfully (and I was supper exited when it worked) but my problems is FS is 
not sending my socket an event when the Leg B is answered.? I get an event only 
when Leg B is hangup.? I must mention, I set filters on the socket to avoid lots 
of the many generated events by FS
Here is my command list on the socket.
?
connect
myevents
filter Event-Name CHANNEL_ANSWER
filter Event-Name CHANNEL_CREATE
filter Event-Name CHANNEL_EXECUTE_COMPLETE
filter Event-Name CHANNEL_BRIDGE
filter Event-Name CHANNEL_UNBRIDGE
filter Event-Name CHANNEL_HANGUP
sendmsg
call-command: execute
execute-app-name: answer
sendmsg
call-command: execute
execute-app-name: sleep
execute-app-arg:1000
event-lock:true
sendmsg
call-command: execute
execute-app-name: play_and_get_digits
execute-app-arg:2 5 3 5000 # ivr/ivr-please_enter_pin_followed_by_pound.wav 
ivr/ivr-please_reenter_your_pin.wav chanDTMF ^\d{5}$
event-lock:true
sendmsg
call-command: execute
execute-app-name: bridge
execute-app-arg:[sip_auth_username=XXXX,sip_auth_password=*******]sofia/internal/123456789 at PROVIDER.COM

event-lock:true
?
Many Thanks,
Ali R.





From lakindia89 at gmail.com  Wed Aug 11 21:24:32 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Thu, 12 Aug 2010 09:54:32 +0530
Subject: [Freeswitch-dev] Socket - bridge answered event
In-Reply-To: <250201.55875.qm@web120608.mail.ne1.yahoo.com>
References: <250201.55875.qm@web120608.mail.ne1.yahoo.com>
Message-ID: <AANLkTimxj-oRveqM9CtYdLSs2ynEbeQ0aqOrxY0z=n_u@mail.gmail.com>

Hi

You register for myevents only.
So you will receive events only for LEG A.
Just try
    events plain all




On Wed, Aug 11, 2010 at 8:00 AM, Ali Roumie <aroumie at yahoo.com> wrote:

> Hello All,
> This is my first post to this list and many thanks to all contributors to
> this
> state of art project. I'm using outbound socket and everything is going
> wonderful with me except one thing so far.
> My logic is simple, collect a PIN from Leg A and once the PIN is
> authorized, I
> bridge the call with a SIP provider. Everything is great, call got bridged
> successfully (and I was supper exited when it worked) but my problems is FS
> is
> not sending my socket an event when the Leg B is answered.  I get an event
> only
> when Leg B is hangup.  I must mention, I set filters on the socket to avoid
> lots
> of the many generated events by FS
> Here is my command list on the socket.
>
> connect
> myevents
> filter Event-Name CHANNEL_ANSWER
> filter Event-Name CHANNEL_CREATE
> filter Event-Name CHANNEL_EXECUTE_COMPLETE
> filter Event-Name CHANNEL_BRIDGE
> filter Event-Name CHANNEL_UNBRIDGE
> filter Event-Name CHANNEL_HANGUP
> sendmsg
> call-command: execute
> execute-app-name: answer
> sendmsg
> call-command: execute
> execute-app-name: sleep
> execute-app-arg:1000
> event-lock:true
> sendmsg
> call-command: execute
> execute-app-name: play_and_get_digits
> execute-app-arg:2 5 3 5000 # ivr/ivr-please_enter_pin_followed_by_pound.wav
> ivr/ivr-please_reenter_your_pin.wav chanDTMF ^\d{5}$
> event-lock:true
> sendmsg
> call-command: execute
> execute-app-name: bridge
>
> execute-app-arg:[sip_auth_username=XXXX,sip_auth_password=*******]sofia/internal/
> 123456789 at PROVIDER.COM
>
> event-lock:true
>
> Many Thanks,
> Ali R.
>
>
>
>
> _______________________________________________
> 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 math.parent at gmail.com  Wed Aug 11 23:28:00 2010
From: math.parent at gmail.com (Mathieu Parent)
Date: Thu, 12 Aug 2010 11:58:00 +0530
Subject: [Freeswitch-dev] Skinny bug
In-Reply-To: <AANLkTinpi8rR5K-ohhfDxQoJ7LAJyt+NHpUtHHJXuH6g@mail.gmail.com>
References: <AANLkTinpi8rR5K-ohhfDxQoJ7LAJyt+NHpUtHHJXuH6g@mail.gmail.com>
Message-ID: <AANLkTikon09kX=HTSSrvzp4AwB+MLWa5YFVi_EkK7+-A@mail.gmail.com>

On Sat, Aug 7, 2010 at 12:33 AM, Sergio Alecha <sergio.alecha at gmail.com> wrote:
>
> I can not make the call transfer module mod_skinny. I am using a cisco 7910
> phone. and FreeSwitch version 1.0.6.

What is your platform? 64bit needs latest git. Also, lot has been done
since 1.0.6.

Can you test again with latest git, and report any success/failure?

Regards

Mathieu Parent


From aroumie at yahoo.com  Thu Aug 12 00:30:23 2010
From: aroumie at yahoo.com (Ali R.)
Date: Thu, 12 Aug 2010 00:30:23 -0700 (PDT)
Subject: [Freeswitch-dev] Socket - bridge answered event
In-Reply-To: <AANLkTimxj-oRveqM9CtYdLSs2ynEbeQ0aqOrxY0z=n_u@mail.gmail.com>
References: <250201.55875.qm@web120608.mail.ne1.yahoo.com>
	<AANLkTimxj-oRveqM9CtYdLSs2ynEbeQ0aqOrxY0z=n_u@mail.gmail.com>
Message-ID: <483598.28088.qm@web120619.mail.ne1.yahoo.com>

Thank you so much for your response!
I have not tried what you suggested but don?t you think sending the command 
"events plain all" on an outbound socket creates lots of?overhead on my app
and the FS event socket module?? 
Correct me if I'm wrong, in outbound socket mode FS spawns a new socket 
connection for each Leg A into my listening server?? Also?if I pass
"events plain all" or the other handy command "events xml all" on each
socket, FS will push the same TCP stream over?all connected sockets.? 
For example, if I got 50 connected Leg A, FS will push the same event?50
times?to my listening?socket?? My app is very sensitive to bandwidth and
that's?the reason I set the filter to just filter out the events that are 
sufficient
for my application logic.
I'm still experiencing with FS and I might be wrong.

P.S: Regarding my issue, I noticed when Leg B is answered the fs_cli logs this
event and that's the one I'm looking for?at my?end point socket.
Thanks,
Ali,



________________________________
From: lakshmanan ganapathy <lakindia89 at gmail.com>
To: freeswitch-dev at lists.freeswitch.org
Sent: Wed, August 11, 2010 9:24:32 PM
Subject: Re: [Freeswitch-dev] Socket - bridge answered event

Hi 

You register for myevents only.
So you will receive events only for LEG A.
Just try 
??? events plain all





On Wed, Aug 11, 2010 at 8:00 AM, Ali Roumie <aroumie at yahoo.com> wrote:

Hello All,
>This is my first post to this list and many thanks to all contributors to this
>state of art project. I'm using outbound socket and everything is going
>wonderful with me except one thing so far.
>My logic is simple, collect a PIN from Leg A and once the PIN is authorized, I
>bridge the call with a SIP provider. Everything is great, call got bridged
>successfully (and I was supper exited when it worked) but my problems is FS is
>not sending my socket an event when the Leg B is answered.? I get an event only
>when Leg B is hangup.? I must mention, I set filters on the socket to avoid 
lots
>of the many generated events by FS
>Here is my command list on the socket.
>?
>connect
>myevents
>filter Event-Name CHANNEL_ANSWER
>filter Event-Name CHANNEL_CREATE
>filter Event-Name CHANNEL_EXECUTE_COMPLETE
>filter Event-Name CHANNEL_BRIDGE
>filter Event-Name CHANNEL_UNBRIDGE
>filter Event-Name CHANNEL_HANGUP
>sendmsg
>call-command: execute
>execute-app-name: answer
>sendmsg
>call-command: execute
>execute-app-name: sleep
>execute-app-arg:1000
>event-lock:true
>sendmsg
>call-command: execute
>execute-app-name: play_and_get_digits
>execute-app-arg:2 5 3 5000 # ivr/ivr-please_enter_pin_followed_by_pound.wav
>ivr/ivr-please_reenter_your_pin.wav chanDTMF ^\d{5}$
>event-lock:true
>sendmsg
>call-command: execute
>execute-app-name: bridge
>execute-app-arg:[sip_auth_username=XXXX,sip_auth_password=*******]sofia/internal/123456789 at PROVIDER.COM
>
>
>
>
>event-lock:true
>?
>Many Thanks,
>Ali R.
>
>
>
>
>_______________________________________________
>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 msc at freeswitch.org  Thu Aug 12 11:10:25 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 12 Aug 2010 11:10:25 -0700
Subject: [Freeswitch-dev] Help needed: review FS book
Message-ID: <AANLkTi=WVzguOHT=d6M38-hFR5OOXp6-rsSY0b_62pt6@mail.gmail.com>

Hello all!

The publishers of the new "Bridge Book" are looking for FS devs/users who
are in a position to write a review. If you would like to write a review
please contact me off list.

Thanks!

-MC
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From aroumie at yahoo.com  Fri Aug 13 00:40:26 2010
From: aroumie at yahoo.com (Ali R.)
Date: Fri, 13 Aug 2010 00:40:26 -0700 (PDT)
Subject: [Freeswitch-dev] Socket - bridge answered event
In-Reply-To: <483598.28088.qm@web120619.mail.ne1.yahoo.com>
References: <250201.55875.qm@web120608.mail.ne1.yahoo.com>
	<AANLkTimxj-oRveqM9CtYdLSs2ynEbeQ0aqOrxY0z=n_u@mail.gmail.com>
	<483598.28088.qm@web120619.mail.ne1.yahoo.com>
Message-ID: <413105.50857.qm@web120611.mail.ne1.yahoo.com>

I just found out that if I use the event socket in inbound mode, I receive the 
CHANNEL_ANSWER event after CHANNEL_BRIDGE which is exactly 
what I'm looking for but in Socket outbound mode.? I'm wondering if?this is by
design or something else.
Many Thanks,
Ali



----- Original Message ----
From: Ali R. <aroumie at yahoo.com>
To: freeswitch-dev at lists.freeswitch.org
Sent: Thu, August 12, 2010 12:30:23 AM
Subject: Re: [Freeswitch-dev] Socket - bridge answered event

Thank you so much for your response!
I have not tried what you suggested but don?t you think sending the command 
"events plain all" on an outbound socket creates lots of?overhead on my app
and the FS event socket module?? 
Correct me if I'm wrong, in outbound socket mode FS spawns a new socket 
connection for each Leg A into my listening server?? Also?if I pass
"events plain all" or the other handy command "events xml all" on each
socket, FS will push the same TCP stream over?all connected sockets.? 
For example, if I got 50 connected Leg A, FS will push the same event?50
times?to my listening?socket?? My app is very sensitive to bandwidth and
that's?the reason I set the filter to just filter out the events that are 
sufficient
for my application logic.
I'm still experiencing with FS and I might be wrong.

P.S: Regarding my issue, I noticed when Leg B is answered the fs_cli logs this
event and that's the one I'm looking for?at my?end point socket.
Thanks,
Ali,



________________________________
From: lakshmanan ganapathy <lakindia89 at gmail.com>
To: freeswitch-dev at lists.freeswitch.org
Sent: Wed, August 11, 2010 9:24:32 PM
Subject: Re: [Freeswitch-dev] Socket - bridge answered event

Hi 

You register for myevents only.
So you will receive events only for LEG A.
Just try 
??? events plain all





On Wed, Aug 11, 2010 at 8:00 AM, Ali Roumie <aroumie at yahoo.com> wrote:

Hello All,
>This is my first post to this list and many thanks to all contributors to this
>state of art project. I'm using outbound socket and everything is going
>wonderful with me except one thing so far.
>My logic is simple, collect a PIN from Leg A and once the PIN is authorized, I
>bridge the call with a SIP provider. Everything is great, call got bridged
>successfully (and I was supper exited when it worked) but my problems is FS is
>not sending my socket an event when the Leg B is answered.? I get an event only
>when Leg B is hangup.? I must mention, I set filters on the socket to avoid 
lots
>of the many generated events by FS
>Here is my command list on the socket.
>?
>connect
>myevents
>filter Event-Name CHANNEL_ANSWER
>filter Event-Name CHANNEL_CREATE
>filter Event-Name CHANNEL_EXECUTE_COMPLETE
>filter Event-Name CHANNEL_BRIDGE
>filter Event-Name CHANNEL_UNBRIDGE
>filter Event-Name CHANNEL_HANGUP
>sendmsg
>call-command: execute
>execute-app-name: answer
>sendmsg
>call-command: execute
>execute-app-name: sleep
>execute-app-arg:1000
>event-lock:true
>sendmsg
>call-command: execute
>execute-app-name: play_and_get_digits
>execute-app-arg:2 5 3 5000 # ivr/ivr-please_enter_pin_followed_by_pound.wav
>ivr/ivr-please_reenter_your_pin.wav chanDTMF ^\d{5}$
>event-lock:true
>sendmsg
>call-command: execute
>execute-app-name: bridge
>execute-app-arg:[sip_auth_username=XXXX,sip_auth_password=*******]sofia/internal/123456789 at PROVIDER.COM
>
>
>
>
>
>event-lock:true
>?
>Many Thanks,
>Ali R.
>
>
>
>
>_______________________________________________
>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
>



? ? ? 


_______________________________________________
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 gmaruzz at celliax.org  Fri Aug 13 14:35:37 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 13 Aug 2010 23:35:37 +0200
Subject: [Freeswitch-dev] mod_skypopen (skype endpoint) changes, please test
Message-ID: <AANLkTinbxLhgv22SmJaKFH2uXabEJ9MyEKMFkMPVCmpU@mail.gmail.com>

Hi FreeSWITCHers,

I've made some long due modifications to mod_skypopen, that maybe
introduced bugs.

Please test with the latest git and report any *new* problem (ok, old
problems too), here in the mailing list, or - much better - in the
Jira.

=========
commit 45c6c4d3e42e3c114b47d52ca2e9fca6b1be8090
Author: Giovanni Maruzzelli <gmaruzz at gmail.com>
Date:   Fri Aug 13 16:19:20 2010 -0500

    skypopen: now answer a call only when directed to do it (before
was trying to answer any incoming call). Lot of changes to a messy
part, so maybe some problem will come out...

    Signed-off-by: Giovanni Maruzzelli <gmaruzz at gmail.com>
===========

Thank you to all,

-giovanni

-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From gmaruzz at celliax.org  Mon Aug 16 09:31:00 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 16 Aug 2010 18:31:00 +0200
Subject: [Freeswitch-dev] mod_skypopen (skype endpoint) changes,
	please test (REPOST)
Message-ID: <AANLkTinM2nd7ruSHLSjigZOAC4GqxUF4Z=VhKaJVD-4=@mail.gmail.com>

Just in case you missed the weekend mailz:
=======

Hi FreeSWITCHers,

I've made some long due modifications to mod_skypopen, that maybe
introduced bugs.

Please test with the latest git and report any *new* problem (ok, old
problems too), here in the mailing list, or - much better - in the
Jira.

=========
commit 45c6c4d3e42e3c114b47d52ca2e9fca6b1be8090
Author: Giovanni Maruzzelli <gmaruzz at gmail.com>
Date:   Fri Aug 13 16:19:20 2010 -0500

   skypopen: now answer a call only when directed to do it (before
was trying to answer any incoming call). Lot of changes to a messy
part, so maybe some problem will come out...

   Signed-off-by: Giovanni Maruzzelli <gmaruzz at gmail.com>
===========

Thank you to all,

-giovanni

-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From lakindia89 at gmail.com  Thu Aug 12 00:44:03 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Thu, 12 Aug 2010 13:14:03 +0530
Subject: [Freeswitch-dev] Socket - bridge answered event
In-Reply-To: <483598.28088.qm@web120619.mail.ne1.yahoo.com>
References: <250201.55875.qm@web120608.mail.ne1.yahoo.com>
	<AANLkTimxj-oRveqM9CtYdLSs2ynEbeQ0aqOrxY0z=n_u@mail.gmail.com>
	<483598.28088.qm@web120619.mail.ne1.yahoo.com>
Message-ID: <AANLkTikH4dsW9Ym8sCe_iqOQEdBBTjj40GJPvXohmg2A@mail.gmail.com>

Ok. I understand your point.

One more way is, when the LEG B answers the call, you will get
CHANNEL_BRIDGE event in the A LEG. There you will have all the informations
regarding the LEG B ( including the UUID of the LEG B ).

May I know What you want to do when LEB B is answered??, so that I can
suggest better ways.



On Thu, Aug 12, 2010 at 1:00 PM, Ali R. <aroumie at yahoo.com> wrote:

> Thank you so much for your response!
> I have not tried what you suggested but don?t you think sending the command
> "events plain all" on an outbound socket creates lots of overhead on my app
> and the FS event socket module?
> Correct me if I'm wrong, in outbound socket mode FS spawns a new socket
> connection for each Leg A into my listening server ? Also if I pass
> "events plain all" or the other handy command "events xml all" on each
> socket, FS will push the same TCP stream over all connected sockets.
> For example, if I got 50 connected Leg A, FS will push the same event 50
> times to my listening socket?  My app is very sensitive to bandwidth and
> that's the reason I set the filter to just filter out the events that are
> sufficient
> for my application logic.
> I'm still experiencing with FS and I might be wrong.
>
> P.S: Regarding my issue, I noticed when Leg B is answered the fs_cli logs
> this
> event and that's the one I'm looking for at my end point socket.
> Thanks,
> Ali,
>
>
>
> ________________________________
> From: lakshmanan ganapathy <lakindia89 at gmail.com>
> To: freeswitch-dev at lists.freeswitch.org
> Sent: Wed, August 11, 2010 9:24:32 PM
> Subject: Re: [Freeswitch-dev] Socket - bridge answered event
>
> Hi
>
> You register for myevents only.
> So you will receive events only for LEG A.
> Just try
>     events plain all
>
>
>
>
>
> On Wed, Aug 11, 2010 at 8:00 AM, Ali Roumie <aroumie at yahoo.com> wrote:
>
> Hello All,
> >This is my first post to this list and many thanks to all contributors to
> this
> >state of art project. I'm using outbound socket and everything is going
> >wonderful with me except one thing so far.
> >My logic is simple, collect a PIN from Leg A and once the PIN is
> authorized, I
> >bridge the call with a SIP provider. Everything is great, call got bridged
> >successfully (and I was supper exited when it worked) but my problems is
> FS is
> >not sending my socket an event when the Leg B is answered.  I get an event
> only
> >when Leg B is hangup.  I must mention, I set filters on the socket to
> avoid
> lots
> >of the many generated events by FS
> >Here is my command list on the socket.
> >
> >connect
> >myevents
> >filter Event-Name CHANNEL_ANSWER
> >filter Event-Name CHANNEL_CREATE
> >filter Event-Name CHANNEL_EXECUTE_COMPLETE
> >filter Event-Name CHANNEL_BRIDGE
> >filter Event-Name CHANNEL_UNBRIDGE
> >filter Event-Name CHANNEL_HANGUP
> >sendmsg
> >call-command: execute
> >execute-app-name: answer
> >sendmsg
> >call-command: execute
> >execute-app-name: sleep
> >execute-app-arg:1000
> >event-lock:true
> >sendmsg
> >call-command: execute
> >execute-app-name: play_and_get_digits
> >execute-app-arg:2 5 3 5000 #
> ivr/ivr-please_enter_pin_followed_by_pound.wav
> >ivr/ivr-please_reenter_your_pin.wav chanDTMF ^\d{5}$
> >event-lock:true
> >sendmsg
> >call-command: execute
> >execute-app-name: bridge
>
> >execute-app-arg:[sip_auth_username=XXXX,sip_auth_password=*******]sofia/internal/
> 123456789 at PROVIDER.COM
> >
> >
> >
> >
> >event-lock:true
> >
> >Many Thanks,
> >Ali R.
> >
> >
> >
> >
> >_______________________________________________
> >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
> >
>
>
>
>
>
>
> _______________________________________________
> 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 di-shi at transnexus.com  Thu Aug 12 18:27:24 2010
From: di-shi at transnexus.com (Di-Shi Sun)
Date: Fri, 13 Aug 2010 09:27:24 +0800
Subject: [Freeswitch-dev] FreeSWITCH mod_osp: undefined symbol
	OSPPTransactionSetSrcNetworkId**
References: <4C641FD7.8090002@laimbock.com>
Message-ID: <F89D7E4B0B304ECC8610469412428163@e520>

Hi Patrick,

I have several questions,
1. Is /usr/lib64 included in the default link search path?
2. Did it report any warning or error message when you compiled mod_osp 
module?
3. Do you install several OSP Toolkit versions?
4. Is there any other libosptk.a under the link search path?
5. How did you build/install OSP Toolkit?

BTW, OSP Toolkit only provides static library. So from the nm output, it 
seems that OSP Toolkit was not linked into mod_osp.so.

Thanks,

Di-Shi Sun.

----- Original Message ----- 
From: "Patrick Laimbock" <patrick at laimbock.com>
To: <di-shi at transnexus.com>
Sent: Friday, August 13, 2010 12:22 AM
Subject: FreeSWITCH mod_osp: undefined symbol 
OSPPTransactionSetSrcNetworkId**


> Dear Mr. Sun,
>
> According to mod_osp.c in the FreeSWITCH sources you are the developer of 
> this module. There is a problem with an undefined symbol in mod_osp with 
> FreeSWITCH git from 8/10:
>
> Platform:   CentOS 5.5 x86_64 (updated)
> FreeSWITCH: git-07b8176 2010-08-10 18-51-06 -0400
> OSPToolkit: 3.6.1
>
> After installing OSPToolkit I enabled mod_osp and built and installed
> FreeSWITCH. When I issue "load mod_osp" I see this error:
>
> 2010-08-11 21:00:29.284157 [CRIT] switch_loadable_module.c:926 Error
> Loading module /opt/freeswitch/mod/mod_osp.so
> **/opt/freeswitch/mod/mod_osp.so: undefined symbol:
> OSPPTransactionSetSrcNetworkId**
>
> I'm out of my league here but I tried to get as much info as possible.
>
> /usr/lib64/osp/libosptk.a has the following symbols:
>
> $ nm /usr/lib64/libosptk.a
> [snip]
> osptransapi.o:
> 00000000000006ed T OSPPTransactionSetSrcNetworkId
> [snip]
>
> Checking mod_osp I see the following:
>
> $ nm /opt/freeswitch/mod/mod_osp.so
> 0000000000209570 d B64CACert
> 0000000000209568 d B64LCert
> 0000000000209560 d B64PKey
>                   U OSPPBase64Decode
>                   U OSPPCleanup
>                   U OSPPInit
>                   U OSPPProviderDelete
>                   U OSPPProviderNew
>                   U OSPPTransactionBuildUsageFromScratch
>                   U OSPPTransactionDelete
>                   U OSPPTransactionGetContext
>                   U OSPPTransactionGetDestProtocol
>                   U OSPPTransactionGetDestinationNetworkId
>                   U OSPPTransactionGetFirstDestination
>                   U OSPPTransactionGetNextDestination
>                   U OSPPTransactionGetNumberPortabilityParameters
>                   U OSPPTransactionGetOperatorName
>                   U OSPPTransactionIsDestOSPEnabled
>                   U OSPPTransactionNew
>                   U OSPPTransactionRecordFailure
>                   U OSPPTransactionReportUsage
>                   U OSPPTransactionRequestAuthorisation
>                   U OSPPTransactionSetCustomInfo
>                   U OSPPTransactionSetDestNetworkId
>                   U OSPPTransactionSetDestinationCount
>                   U OSPPTransactionSetDiversion
>                   U OSPPTransactionSetForwardCodec
>                   U OSPPTransactionSetNetworkIds
>                   U OSPPTransactionSetNumberPortability
>                   U OSPPTransactionSetOctets
>                   U OSPPTransactionSetPackets
>                   U OSPPTransactionSetReverseCodec
>                   U OSPPTransactionSetServiceType
>                   U OSPPTransactionSetSrcNetworkId  <--
> [snip]
>
> According to the nm manpage, the "U" means "undefined" so mod_osp does
> not seem to pick up libosptk.a (during linking?).
>
> Would you perhaps have any suggestions how I can fix this error?
>
> Please let me know if you need more information.
>
> Many thanks in advance!
>
> Kind regards,
> Patrick Laimbock
>
> 



From di-shi at transnexus.com  Fri Aug 13 15:56:37 2010
From: di-shi at transnexus.com (Di-Shi Sun)
Date: Sat, 14 Aug 2010 06:56:37 +0800
Subject: [Freeswitch-dev] FreeSWITCH mod_osp: undefined symbol
	OSPPTransactionSetSrcNetworkId**
References: <4C641FD7.8090002@laimbock.com>
	<F89D7E4B0B304ECC8610469412428163@e520>
	<4C64EF3C.9060901@laimbock.com>
	<DEE6D1CFC8B148E29F7B42A0C747CC2B@e520>
	<4C6574CB.8050504@laimbock.com>
Message-ID: <73B004C6B98A409EA49BB600760E21B8@e520>

Hi Patrick,

It appears that you skipped one necessary building step. Please see the doc 
I sent to you, section 3.1, step 4. It makes FreeRADIUS to generate correct 
Makefile.in/Makefile for mod_osp.

Regards,

Di-Shi Sun.
----- Original Message ----- 
From: "Patrick Laimbock" <patrick at laimbock.com>
To: "Di-Shi Sun" <di-shi at transnexus.com>
Sent: Saturday, August 14, 2010 12:37 AM
Subject: Re: FreeSWITCH mod_osp: undefined symbol 
OSPPTransactionSetSrcNetworkId**


> Hi Di-Shi,
>
> Files attached with the exception of:
>
> 8. src/mod/applications/mod_osp/Makefile.in <-- does not exist but
> Makefile.am does so I have included that one.
>
> 10. src/mod/applications/mod_osp/mod_osp.log <-- does not exist
>
> Best regards,
> Patrick
>
>
> On 08/13/2010 10:06 AM, Di-Shi Sun wrote:
>> Hi Patrick,
>>
>> Would you please send me the following files?
>>
>> 1. /usr/include/osp/osplibversion.h
>> 2. /home/patrick/src/freeswitch/build/modules.conf.in
>> 3. /home/patrick/src/freeswitch/modules.conf
>> 4. /home/patrick/src/freeswitch/configure.in
>> 5. /home/patrick/src/freeswitch/configure
>> 6. /home/patrick/src/freeswitch/config.log
>> 7. /home/patrick/src/freeswitch/config.status
>> 8. /home/patrick/src/freeswitch/src/mod/applications/mod_osp/Makefile.in
>> 9. /home/patrick/src/freeswitch/src/mod/applications/mod_osp/Makefile
>> 10. /home/patrick/src/freeswitch/src/mod/applications/mod_osp/mod_osp.log
>>
>> Thanks,
>>
>> Di-Shi Sun.
>>
>> ----- Original Message ----- From: "Patrick Laimbock"
>> <patrick at laimbock.com>
>> To: "Di-Shi Sun" <di-shi at transnexus.com>
>> Sent: Friday, August 13, 2010 3:07 PM
>> Subject: Re: FreeSWITCH mod_osp: undefined symbol
>> OSPPTransactionSetSrcNetworkId**
>>
>>
>>> Hi Di-Shi,
>>>
>>> Thank you for your feedback. Comments inline.
>>>
>>> On 08/13/2010 03:27 AM, Di-Shi Sun wrote:
>>>> Hi Patrick,
>>>>
>>>> I have several questions,
>>>> 1. Is /usr/lib64 included in the default link search path?
>>>
>>> Not sure what you mean with the default link search path. Can you
>>> please explain?
>>>
>>>> 2. Did it report any warning or error message when you compiled mod_osp
>>>> module?
>>>
>>> No, see output:
>>>
>>> Creating mod_enum.la
>>>
>>> making all mod_osp
>>> /home/patrick/src/freeswitch/src/mod/applications/mod_osp
>>> Compiling
>>> /home/patrick/src/freeswitch/src/mod/applications/mod_osp/mod_osp.c...
>>> mkdir .libs
>>> Compiling
>>> /home/patrick/src/freeswitch/src/mod/applications/mod_osp/mod_osp.c ...
>>> Creating mod_osp.so...
>>>
>>> making all mod_fifo
>>>
>>>
>>>> 3. Do you install several OSP Toolkit versions?
>>>
>>> No, only 3.6.1
>>>
>>>> 4. Is there any other libosptk.a under the link search path?
>>>
>>> No
>>>
>>> # find / -name libosptk.a -type f
>>> /usr/lib64/libosptk.a
>>> #
>>>
>>>> 5. How did you build/install OSP Toolkit?
>>>
>>> Completely according to the guide you sent me (thanks by the way)
>>> including openssl-0.9.8i as recommended in the guide.
>>>
>>>> BTW, OSP Toolkit only provides static library. So from the nm output, 
>>>> it
>>>> seems that OSP Toolkit was not linked into mod_osp.so.
>>>
>>> Agree. I looked at the Makefiles of other modules but I do not know
>>> anything about libtool and the auto* tools so am out of my league. My
>>> guess is something may need to be added to Makefile.am but I don't
>>> know what. If you have any ideas then I'm happy to test them.
>>>
>>> Thanks,
>>> Patrick
>>>
>>
>
> 



From di-shi at transnexus.com  Sat Aug 14 18:16:14 2010
From: di-shi at transnexus.com (Di-Shi Sun)
Date: Sun, 15 Aug 2010 09:16:14 +0800
Subject: [Freeswitch-dev] FreeSWITCH mod_osp: undefined symbol
	OSPPTransactionSetSrcNetworkId**
References: <4C641FD7.8090002@laimbock.com>
	<F89D7E4B0B304ECC8610469412428163@e520>
	<4C64EF3C.9060901@laimbock.com>
	<DEE6D1CFC8B148E29F7B42A0C747CC2B@e520>
	<4C6574CB.8050504@laimbock.com>
	<73B004C6B98A409EA49BB600760E21B8@e520>
	<4C66D3D9.3010003@laimbock.com>
Message-ID: <25C2C58418D04DE889C1F2FC7C43A001@e520>

Hi Patrick,

Would you please open the doc I sent by MS Word? OpenOffice is good but may 
not handle this doc correctly. In fact, the section I mentioned shows up as 
section 1.5 Build FreeSWITCH with OSP module, step 4 Change configure.in to 
create the OSP module Makefile in the pdf you attached.

BTW, sorry for the typo of FreeRADIUS. :) I am working on another project.

Regards,

Di-Shi Sun.

----- Original Message ----- 
From: "Patrick Laimbock" <patrick at laimbock.com>
To: "Di-Shi Sun" <di-shi at transnexus.com>
Sent: Sunday, August 15, 2010 1:35 AM
Subject: Re: FreeSWITCH mod_osp: undefined symbol 
OSPPTransactionSetSrcNetworkId**


> Hi Di-Shi,
>
> Thank you for your feedback. I searched the doc you sent me but there is
> no section 3.1. step 4 (I opened the doc with OpenOffice.org). There is
> also no mentioning of FreeRADIUS in the entire document. The doc you
> sent me is titled "FreeSWITCH OSP Module User Guide July 8, 2010". Is
> that the correct version? I have attached a pdf version of the doc.
>
> Have a good weekend!
>
> Best regards,
> Patrick
>
> On 08/14/2010 12:56 AM, Di-Shi Sun wrote:
>> Hi Patrick,
>>
>> It appears that you skipped one necessary building step. Please see the
>> doc I sent to you, section 3.1, step 4. It makes FreeRADIUS to generate
>> correct Makefile.in/Makefile for mod_osp.
>>
>> Regards,
>>
>> Di-Shi Sun.
>> ----- Original Message ----- From: "Patrick Laimbock"
>> <patrick at laimbock.com>
>> To: "Di-Shi Sun" <di-shi at transnexus.com>
>> Sent: Saturday, August 14, 2010 12:37 AM
>> Subject: Re: FreeSWITCH mod_osp: undefined symbol
>> OSPPTransactionSetSrcNetworkId**
>>
>>
>>> Hi Di-Shi,
>>>
>>> Files attached with the exception of:
>>>
>>> 8. src/mod/applications/mod_osp/Makefile.in <-- does not exist but
>>> Makefile.am does so I have included that one.
>>>
>>> 10. src/mod/applications/mod_osp/mod_osp.log <-- does not exist
>>>
>>> Best regards,
>>> Patrick
>>>
>>>
>>> On 08/13/2010 10:06 AM, Di-Shi Sun wrote:
>>>> Hi Patrick,
>>>>
>>>> Would you please send me the following files?
>>>>
>>>> 1. /usr/include/osp/osplibversion.h
>>>> 2. /home/patrick/src/freeswitch/build/modules.conf.in
>>>> 3. /home/patrick/src/freeswitch/modules.conf
>>>> 4. /home/patrick/src/freeswitch/configure.in
>>>> 5. /home/patrick/src/freeswitch/configure
>>>> 6. /home/patrick/src/freeswitch/config.log
>>>> 7. /home/patrick/src/freeswitch/config.status
>>>> 8. 
>>>> /home/patrick/src/freeswitch/src/mod/applications/mod_osp/Makefile.in
>>>> 9. /home/patrick/src/freeswitch/src/mod/applications/mod_osp/Makefile
>>>> 10.
>>>> /home/patrick/src/freeswitch/src/mod/applications/mod_osp/mod_osp.log
>>>>
>>>> Thanks,
>>>>
>>>> Di-Shi Sun.
>>>>
>>>> ----- Original Message ----- From: "Patrick Laimbock"
>>>> <patrick at laimbock.com>
>>>> To: "Di-Shi Sun" <di-shi at transnexus.com>
>>>> Sent: Friday, August 13, 2010 3:07 PM
>>>> Subject: Re: FreeSWITCH mod_osp: undefined symbol
>>>> OSPPTransactionSetSrcNetworkId**
>>>>
>>>>
>>>>> Hi Di-Shi,
>>>>>
>>>>> Thank you for your feedback. Comments inline.
>>>>>
>>>>> On 08/13/2010 03:27 AM, Di-Shi Sun wrote:
>>>>>> Hi Patrick,
>>>>>>
>>>>>> I have several questions,
>>>>>> 1. Is /usr/lib64 included in the default link search path?
>>>>>
>>>>> Not sure what you mean with the default link search path. Can you
>>>>> please explain?
>>>>>
>>>>>> 2. Did it report any warning or error message when you compiled
>>>>>> mod_osp
>>>>>> module?
>>>>>
>>>>> No, see output:
>>>>>
>>>>> Creating mod_enum.la
>>>>>
>>>>> making all mod_osp
>>>>> /home/patrick/src/freeswitch/src/mod/applications/mod_osp
>>>>> Compiling
>>>>> /home/patrick/src/freeswitch/src/mod/applications/mod_osp/mod_osp.c...
>>>>> mkdir .libs
>>>>> Compiling
>>>>> /home/patrick/src/freeswitch/src/mod/applications/mod_osp/mod_osp.c 
>>>>> ...
>>>>> Creating mod_osp.so...
>>>>>
>>>>> making all mod_fifo
>>>>>
>>>>>
>>>>>> 3. Do you install several OSP Toolkit versions?
>>>>>
>>>>> No, only 3.6.1
>>>>>
>>>>>> 4. Is there any other libosptk.a under the link search path?
>>>>>
>>>>> No
>>>>>
>>>>> # find / -name libosptk.a -type f
>>>>> /usr/lib64/libosptk.a
>>>>> #
>>>>>
>>>>>> 5. How did you build/install OSP Toolkit?
>>>>>
>>>>> Completely according to the guide you sent me (thanks by the way)
>>>>> including openssl-0.9.8i as recommended in the guide.
>>>>>
>>>>>> BTW, OSP Toolkit only provides static library. So from the nm
>>>>>> output, it
>>>>>> seems that OSP Toolkit was not linked into mod_osp.so.
>>>>>
>>>>> Agree. I looked at the Makefiles of other modules but I do not know
>>>>> anything about libtool and the auto* tools so am out of my league. My
>>>>> guess is something may need to be added to Makefile.am but I don't
>>>>> know what. If you have any ideas then I'm happy to test them.
>>>>>
>>>>> Thanks,
>>>>> Patrick
>>>>>
>>>>
>>>
>>>
>>
>
> 



From ari.siitonen at voicestream.fi  Mon Aug 16 00:06:56 2010
From: ari.siitonen at voicestream.fi (Ari Siitonen)
Date: Mon, 16 Aug 2010 10:06:56 +0300
Subject: [Freeswitch-dev] New Hardware for Freeswitch
Message-ID: <69C8106D-1944-41F6-ABE7-C5BF1A2C7675@voicestream.fi>

Hello all!

I would like to announce a new hardware development project:

- ISDN Pri interface card 1-8 channels with/without QUICC for ISDN  
stack + dsp
- 1-8 POTS line card
- ISDN Bri line card network/user side

Our company has manufactured such cards and IVR-server back in last  
century,
and we would like to make the "new generation" cards compatible with
freeswitch.

So, to start Hardware redevelopment in right direction, any pointers and
suggestions on driver design would be greatly appreciated. And if the
existing cards lack some features, we will try to incorporate them (hw
conference bridging?)

Also, pilot users will be given free cards to test and evaluate.



Best Regards,

Ari Siitonen
Voice Stream Oy
Finland


From techmicroncom at yahoo.com  Mon Aug 16 09:15:23 2010
From: techmicroncom at yahoo.com (Tech Micron)
Date: Mon, 16 Aug 2010 09:15:23 -0700 (PDT)
Subject: [Freeswitch-dev] Sip RFC2543 and FreeSwitch
Message-ID: <982116.87512.qm@web120604.mail.ne1.yahoo.com>

Hi everyone,

I have a SIP device that is working with RFC 2543. I am trying to send call to 
FreeSwitch, but none of my calls are going through.
As I activated the debug on SOFIA, it returns an error as below. Can anyone help 
me to solve the problem? any hints?


recv 660 bytes from udp/[66.220.15.230]:5060 at 18:45:21.239029:
   ------------------------------------------------------------------------
   INVITE sip:1000 at 66.220.15.234 SIP/2.0
   v: SIP/2.0/UDP 66.220.15.230
   To: sip:1000 at 66.220.15.234 <sip:1000 at 66.220.15.234>
   From: john odu 
<sip:26679778 at 66.220.15.230>;tag=4c5c58410003ac2076f6302c11fc3b7b
   i: 41585c4c2ba3030035e697f0a586dbaa at 66.220.15.230
   CSeq: 1 INVITE
   m: john odu <sip:26679778 at 66.220.15.230>
   Supported: com.lilo.conferencing, com.lilo.mux
   User-Agent: liloSipMCU/3.1.6.200801281402
   c: application/sdp
   l: 212
   
   v=0
   o=mcu 0 0 IN IP4 66.220.15.230
   s=sip:26679778 at 66.220.15.230
   c=IN IP4 66.220.15.230
   t=0 0
   m=audio 2006 RTP/AVP 0 111
   a=ptime:20
   a=rtpmap:0 PCMU/8000
   a=rtpmap:111 telephone-event/8000
   a=xlilossrc:8
   ------------------------------------------------------------------------
tport_deliver(0x97a1fb0): msg 0xb762bef8 (660 bytes) from 
udp/66.220.15.230:5060/sip next=(nil)
nta: received INVITE sip:1000 at 66.220.15.234 SIP/2.0 (CSeq 1)
nta: INVITE has bad To header
nta: INVITE (1) is Bad To Header
tport_tsend(0x97a1fb0) tpn = UDP/66.220.15.230:5060
tport_resolve addrinfo = 66.220.15.230:5060
tport_by_addrinfo(0x97a1fb0): not found by name UDP/66.220.15.230:5060
tport_vsend(0x97a1fb0): 239 bytes of 239 to udp/66.220.15.230:5060
tport_vsend returned 239
send 239 bytes to udp/[66.220.15.230]:5060 at 18:45:21.239369:
   ------------------------------------------------------------------------
   SIP/2.0 400 Bad To Header
   Via: SIP/2.0/UDP 66.220.15.230
   From: john odu 
<sip:26679778 at 66.220.15.230>;tag=4c5c58410003ac2076f6302c11fc3b7b
   Call-ID: 41585c4c2ba3030035e697f0a586dbaa at 66.220.15.230
   CSeq: 1 INVITE
   Content-Length: 0


      
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From brian at freeswitch.org  Mon Aug 16 09:51:05 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 16 Aug 2010 11:51:05 -0500
Subject: [Freeswitch-dev] Sip RFC2543 and FreeSwitch
In-Reply-To: <982116.87512.qm@web120604.mail.ne1.yahoo.com>
References: <982116.87512.qm@web120604.mail.ne1.yahoo.com>
Message-ID: <14048754-5872-4E7F-9C16-4164E3104BE8@freeswitch.org>

I'm pretty sure this to header is 100% invalid.

/b

On Aug 16, 2010, at 11:15 AM, Tech Micron wrote:

>    To: sip:1000 at 66.220.15.234 <sip:1000 at 66.220.15.234>



From techmicroncom at yahoo.com  Mon Aug 16 09:54:32 2010
From: techmicroncom at yahoo.com (Tech Micron)
Date: Mon, 16 Aug 2010 09:54:32 -0700 (PDT)
Subject: [Freeswitch-dev] FreeSwitch - SIP RFC 2543
Message-ID: <692542.19436.qm@web120604.mail.ne1.yahoo.com>

Hi everyone,

I have a SIP device that is working with RFC 2543. I  am trying to send call 
using FreeSwitch (Sofia-SIP), but none of my calls are going  through.
I have traced the debug on SOFIA and it returned an error as below. Can anyone 
help me to solve the problem? any hints?


recv 660 bytes from udp/[66.220.15.230]:5060 at 18:45:21.239029:
   ------------------------------------------------------------------------
   INVITE sip:1000 at 66.220.15.234 SIP/2.0
   v: SIP/2.0/UDP 66.220.15.230
   To: sip:1000 at 66.220.15.234 <sip:1000 at 66.220.15.234>
   From: john odu 
<sip:26679778 at 66.220.15.230>;tag=4c5c58410003ac2076f6302c11fc3b7b
   i: 41585c4c2ba3030035e697f0a586dbaa at 66.220.15.230
   CSeq: 1  INVITE
   m: john odu <sip:26679778 at 66.220.15.230>
   Supported: com.lilo.conferencing, com.lilo.mux
   User-Agent: liloSipMCU/3.1.6.200801281402
   c: application/sdp
   l: 212
   
   v=0
   o=mcu 0 0 IN IP4 66.220.15.230
   s=sip:26679778 at 66.220.15.230
   c=IN IP4 66.220.15.230
   t=0 0
   m=audio 2006 RTP/AVP 0 111
   a=ptime:20
   a=rtpmap:0 PCMU/8000
   a=rtpmap:111 telephone-event/8000
   a=xlilossrc:8
   ------------------------------------------------------------------------
tport_deliver(0x97a1fb0): msg 0xb762bef8 (660 bytes) from 
udp/66.220.15.230:5060/sip next=(nil)
nta: received INVITE sip:1000 at 66.220.15.234 SIP/2.0 (CSeq 1)
nta: INVITE has bad To  header
nta: INVITE (1) is Bad To Header
tport_tsend(0x97a1fb0) tpn = UDP/66.220.15.230:5060
tport_resolve addrinfo = 66.220.15.230:5060
tport_by_addrinfo(0x97a1fb0): not found by name UDP/66.220.15.230:5060
tport_vsend(0x97a1fb0): 239 bytes of 239 to udp/66.220.15.230:5060
tport_vsend returned 239
send 239 bytes to udp/[66.220.15.230]:5060 at 18:45:21.239369:
   ------------------------------------------------------------------------
   SIP/2.0 400 Bad To Header
   Via: SIP/2.0/UDP 66.220.15.230
   From: john odu 
<sip:26679778 at 66.220.15.230>;tag=4c5c58410003ac2076f6302c11fc3b7b
   Call-ID: 41585c4c2ba3030035e697f0a586dbaa at 66.220.15.230
   CSeq: 1 INVITE
   Content-Length: 0



      
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From brian at freeswitch.org  Mon Aug 16 10:00:41 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 16 Aug 2010 12:00:41 -0500
Subject: [Freeswitch-dev] FreeSwitch - SIP RFC 2543
In-Reply-To: <692542.19436.qm@web120604.mail.ne1.yahoo.com>
References: <692542.19436.qm@web120604.mail.ne1.yahoo.com>
Message-ID: <C4627B01-7EDD-4FD6-9E91-4B1733596743@freeswitch.org>

Please pick exactly ONE list not TWO... I already answered you the obvious issue is the to header your device is sending is invalid.

/b

On Aug 16, 2010, at 11:54 AM, Tech Micron wrote:

> Hi everyone,
> 
> I have a SIP device that is working with RFC 2543. I am trying to send call using FreeSwitch (Sofia-SIP), but none of my calls are going through.
> I have traced the debug on SOFIA and it returned an error as below. Can anyone help me to solve the problem? any hints?



From anthony.minessale at gmail.com  Mon Aug 16 10:15:32 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 16 Aug 2010 12:15:32 -0500
Subject: [Freeswitch-dev] Sip RFC2543 and FreeSwitch
In-Reply-To: <14048754-5872-4E7F-9C16-4164E3104BE8@freeswitch.org>
References: <982116.87512.qm@web120604.mail.ne1.yahoo.com>
	<14048754-5872-4E7F-9C16-4164E3104BE8@freeswitch.org>
Message-ID: <AANLkTi=9u0iqm4mXnNgwDkCM5fq=B33hFitRomGHXUhz@mail.gmail.com>

everything outside the <> should be in ""


On Mon, Aug 16, 2010 at 11:51 AM, Brian West <brian at freeswitch.org> wrote:
> I'm pretty sure this to header is 100% invalid.
>
> /b
>
> On Aug 16, 2010, at 11:15 AM, Tech Micron wrote:
>
>> ? ?To: sip:1000 at 66.220.15.234 <sip:1000 at 66.220.15.234>
>
>
> _______________________________________________
> 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 mike at jerris.com  Mon Aug 16 10:24:59 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 16 Aug 2010 13:24:59 -0400
Subject: [Freeswitch-dev] Sip RFC2543 and FreeSwitch
In-Reply-To: <982116.87512.qm@web120604.mail.ne1.yahoo.com>
References: <982116.87512.qm@web120604.mail.ne1.yahoo.com>
Message-ID: <F88358D3-E526-418F-B2CF-67CC6D6A9269@jerris.com>

If you send to more mailing lists, it doesn't mean you will get a different answer.

Mike

On Aug 16, 2010, at 12:15 PM, Tech Micron wrote:

> Hi everyone,
> 
> I have a SIP device that is working with RFC 2543. I am trying to send call to FreeSwitch, but none of my calls are going through.
> As I activated the debug on SOFIA, it returns an error as below. Can anyone help me to solve the problem? any hints?
> 
> 
> recv 660 bytes from udp/[66.220.15.230]:5060 at 18:45:21.239029:
>    ------------------------------------------------------------------------
>    INVITE sip:1000 at 66.220.15.234 SIP/2.0
>    v: SIP/2.0/UDP 66.220.15.230
>    To: sip:1000 at 66.220.15.234 <sip:1000 at 66.220.15.234>
>    From: john odu <sip:26679778 at 66.220.15.230>;tag=4c5c58410003ac2076f6302c11fc3b7b
>    i: 41585c4c2ba3030035e697f0a586dbaa at 66.220.15.230
>    CSeq: 1 INVITE
>    m: john odu <sip:26679778 at 66.220.15.230>
>    Supported: com.lilo.conferencing, com.lilo.mux
>    User-Agent: liloSipMCU/3.1.6.200801281402
>    c: application/sdp
>    l: 212
>    
>    v=0
>    o=mcu 0 0 IN IP4 66.220.15.230
>    s=sip:26679778 at 66.220.15.230
>    c=IN IP4 66.220.15.230
>    t=0 0
>    m=audio 2006 RTP/AVP 0 111
>    a=ptime:20
>    a=rtpmap:0 PCMU/8000
>    a=rtpmap:111 telephone-event/8000
>    a=xlilossrc:8
>    ------------------------------------------------------------------------
> tport_deliver(0x97a1fb0): msg 0xb762bef8 (660 bytes) from udp/66.220.15.230:5060/sip next=(nil)
> nta: received INVITE sip:1000 at 66.220.15.234 SIP/2.0 (CSeq 1)
> nta: INVITE has bad To header
> nta: INVITE (1) is Bad To Header
> tport_tsend(0x97a1fb0) tpn = UDP/66.220.15.230:5060
> tport_resolve addrinfo = 66.220.15.230:5060
> tport_by_addrinfo(0x97a1fb0): not found by name UDP/66.220.15.230:5060
> tport_vsend(0x97a1fb0): 239 bytes of 239 to udp/66.220.15.230:5060
> tport_vsend returned 239
> send 239 bytes to udp/[66.220.15.230]:5060 at 18:45:21.239369:
>    ------------------------------------------------------------------------
>    SIP/2.0 400 Bad To Header
>    Via: SIP/2.0/UDP 66.220.15.230
>    From: john odu <sip:26679778 at 66.220.15.230>;tag=4c5c58410003ac2076f6302c11fc3b7b
>    Call-ID: 41585c4c2ba3030035e697f0a586dbaa at 66.220.15.230
>    CSeq: 1 INVITE
>    Content-Length: 0

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From techmicroncom at yahoo.com  Mon Aug 16 10:33:21 2010
From: techmicroncom at yahoo.com (Tech Micron)
Date: Mon, 16 Aug 2010 10:33:21 -0700 (PDT)
Subject: [Freeswitch-dev] Sip RFC2543 and FreeSwitch
In-Reply-To: <AANLkTi=9u0iqm4mXnNgwDkCM5fq=B33hFitRomGHXUhz@mail.gmail.com>
References: <982116.87512.qm@web120604.mail.ne1.yahoo.com>
	<14048754-5872-4E7F-9C16-4164E3104BE8@freeswitch.org>
	<AANLkTi=9u0iqm4mXnNgwDkCM5fq=B33hFitRomGHXUhz@mail.gmail.com>
Message-ID: <657065.19228.qm@web120601.mail.ne1.yahoo.com>

Hi Anthony,

According to RFC3261 and RFC 2543, Double quote "" is not mandatory. But I have 
seen it in some debug traces from other SoftPhones.

Ref: Section 20.39 of RFC 3261 
Ref: Section 6.37 RFC 2543

The following are examples of valid To headers:
To: The Operator <sip:operator at cs.columbia.edu>;tag=287447
To: sip:+12125551212 at server.phone2net.com


John



________________________________
From: Anthony Minessale <anthony.minessale at gmail.com>
To: freeswitch-dev at lists.freeswitch.org
Sent: Mon, August 16, 2010 1:15:32 PM
Subject: Re: [Freeswitch-dev] Sip RFC2543 and FreeSwitch

everything outside the <> should be in ""


On Mon, Aug 16, 2010 at 11:51 AM, Brian West <brian at freeswitch.org> wrote:
> I'm pretty sure this to header is 100% invalid.
>
> /b
>
> On Aug 16, 2010, at 11:15 AM, Tech Micron wrote:
>
>>    To: sip:1000 at 66.220.15.234 <sip:1000 at 66.220.15.234>
>
>
> _______________________________________________
> 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

_______________________________________________
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 brian at freeswitch.org  Mon Aug 16 10:41:38 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 16 Aug 2010 12:41:38 -0500
Subject: [Freeswitch-dev] Sip RFC2543 and FreeSwitch
In-Reply-To: <657065.19228.qm@web120601.mail.ne1.yahoo.com>
References: <982116.87512.qm@web120604.mail.ne1.yahoo.com>
	<14048754-5872-4E7F-9C16-4164E3104BE8@freeswitch.org>
	<AANLkTi=9u0iqm4mXnNgwDkCM5fq=B33hFitRomGHXUhz@mail.gmail.com>
	<657065.19228.qm@web120601.mail.ne1.yahoo.com>
Message-ID: <D57EEA86-779B-44C2-9D8C-F030E6AC036F@freeswitch.org>

It sure is if you have an @ sign... or you need to URL encode the value listed.

/b

On Aug 16, 2010, at 12:33 PM, Tech Micron wrote:

> Hi Anthony,
> 
> According to RFC3261 and RFC 2543, Double quote "" is not mandatory. But I have seen it in some debug traces from other SoftPhones.
> 
> Ref: Section 20.39 of RFC 3261 
> Ref: Section 6.37 RFC 2543
> 
> The following are examples of valid To headers:
> To: The Operator <sip:operator at cs.columbia.edu>;tag=287447
> To: sip:+12125551212 at server.phone2net.com
> 
> John



From herrold at owlriver.com  Mon Aug 16 12:59:32 2010
From: herrold at owlriver.com (R P Herrold)
Date: Mon, 16 Aug 2010 15:59:32 -0400 (EDT)
Subject: [Freeswitch-dev] Sip RFC2543 and FreeSwitch
In-Reply-To: <657065.19228.qm@web120601.mail.ne1.yahoo.com>
References: <982116.87512.qm@web120604.mail.ne1.yahoo.com>
	<14048754-5872-4E7F-9C16-4164E3104BE8@freeswitch.org>
	<AANLkTi=9u0iqm4mXnNgwDkCM5fq=B33hFitRomGHXUhz@mail.gmail.com>
	<657065.19228.qm@web120601.mail.ne1.yahoo.com>
Message-ID: <alpine.LRH.1.00.1008161552100.27298@oebafba.bjyevire.pbz>

On Mon, 16 Aug 2010, Tech Micron wrote:

> Hi Anthony,
>
> According to RFC3261 and RFC 2543, Double quote "" is not 
> mandatory. But I have seen it in some debug traces from 
> other SoftPhones.
>
> Ref: Section 20.39 of RFC 3261

I am no RFC lawyer, but inter alia, that out refers to RFC 
2616, and the relevant part is the description of 'tokens' in 
a 'field', and when certain separaters are present, must be 
protected in a QUOTE form

http://www.ietf.org/rfc/rfc2616.txt at section 2.2

   Many HTTP/1.1 header field values consist of words separated by LWS
    or special characters. These special characters MUST be in a quoted
    string to be used within a parameter value (as defined in section
    3.6).

        token          = 1*<any CHAR except CTLs or separators>
        separators     = "(" | ")" | "<" | ">" | "@"
                       | "," | ";" | ":" | "\" | <">
                       | "/" | "[" | "]" | "?" | "="
                       | "{" | "}" | SP | HT

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

Your target string as indicated by the error message in 
the first post was:

 	To: sip:1000 at 66.220.15.234 <sip:1000 at 66.220.15.234>

and the relevant unprotected section:
 	sip:1000 at 66.220.15.234
contains: ":" and "@"
without such quotation.  I see Brian mentioned one, but it 
looks as though two are required to be protected

A proper line might look like:

 	To: "sip:1000 at 66.220.15.234" <sip:1000 at 66.220.15.234>

-- Russ herrold


From brian at freeswitch.org  Mon Aug 16 13:06:26 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 16 Aug 2010 15:06:26 -0500
Subject: [Freeswitch-dev] Sip RFC2543 and FreeSwitch
In-Reply-To: <alpine.LRH.1.00.1008161552100.27298@oebafba.bjyevire.pbz>
References: <982116.87512.qm@web120604.mail.ne1.yahoo.com>
	<14048754-5872-4E7F-9C16-4164E3104BE8@freeswitch.org>
	<AANLkTi=9u0iqm4mXnNgwDkCM5fq=B33hFitRomGHXUhz@mail.gmail.com>
	<657065.19228.qm@web120601.mail.ne1.yahoo.com>
	<alpine.LRH.1.00.1008161552100.27298@oebafba.bjyevire.pbz>
Message-ID: <7F272F1C-2372-4C69-BD6A-1024E9DD9810@freeswitch.org>

Russ,
	I love when someone steps up to help out like you have... Bravo.

Thanks,
Brian

On Aug 16, 2010, at 2:59 PM, R P Herrold wrote:

> I am no RFC lawyer, but inter alia, that out refers to RFC 
> 2616, and the relevant part is the description of 'tokens' in 
> a 'field', and when certain separaters are present, must be 
> protected in a QUOTE form
> 
> http://www.ietf.org/rfc/rfc2616.txt at section 2.2
> 
>   Many HTTP/1.1 header field values consist of words separated by LWS
>    or special characters. These special characters MUST be in a quoted
>    string to be used within a parameter value (as defined in section
>    3.6).
> 
>        token          = 1*<any CHAR except CTLs or separators>
>        separators     = "(" | ")" | "<" | ">" | "@"
>                       | "," | ";" | ":" | "\" | <">
>                       | "/" | "[" | "]" | "?" | "="
>                       | "{" | "}" | SP | HT
> 
> ---------------------------
> 
> Your target string as indicated by the error message in 
> the first post was:
> 
> 	To: sip:1000 at 66.220.15.234 <sip:1000 at 66.220.15.234>
> 
> and the relevant unprotected section:
> 	sip:1000 at 66.220.15.234
> contains: ":" and "@"
> without such quotation.  I see Brian mentioned one, but it 
> looks as though two are required to be protected
> 
> A proper line might look like:
> 
> 	To: "sip:1000 at 66.220.15.234" <sip:1000 at 66.220.15.234>
> 
> -- Russ herrold



From msc at freeswitch.org  Mon Aug 16 13:10:02 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 16 Aug 2010 13:10:02 -0700
Subject: [Freeswitch-dev] Sip RFC2543 and FreeSwitch
In-Reply-To: <alpine.LRH.1.00.1008161552100.27298@oebafba.bjyevire.pbz>
References: <982116.87512.qm@web120604.mail.ne1.yahoo.com>
	<14048754-5872-4E7F-9C16-4164E3104BE8@freeswitch.org>
	<AANLkTi=9u0iqm4mXnNgwDkCM5fq=B33hFitRomGHXUhz@mail.gmail.com>
	<657065.19228.qm@web120601.mail.ne1.yahoo.com>
	<alpine.LRH.1.00.1008161552100.27298@oebafba.bjyevire.pbz>
Message-ID: <AANLkTikeVYNzZufa_ewceMaDs10h6b3za4E3kRTuN4_T@mail.gmail.com>

On Mon, Aug 16, 2010 at 12:59 PM, R P Herrold <herrold at owlriver.com> wrote:

> On Mon, 16 Aug 2010, Tech Micron wrote:
>
> > Hi Anthony,
> >
> > According to RFC3261 and RFC 2543, Double quote "" is not
> > mandatory. But I have seen it in some debug traces from
> > other SoftPhones.
> >
> > Ref: Section 20.39 of RFC 3261
>
> I am no RFC lawyer, but inter alia, that out refers to RFC
>

Well, you may not be an RFC lawyer but you sure play one well on this list!
:)
-MC
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From jan.berger at video24.no  Mon Aug 16 14:28:19 2010
From: jan.berger at video24.no (Jan Berger)
Date: Mon, 16 Aug 2010 23:28:19 +0200
Subject: [Freeswitch-dev] New Hardware for Freeswitch
In-Reply-To: <69C8106D-1944-41F6-ABE7-C5BF1A2C7675@voicestream.fi>
References: <69C8106D-1944-41F6-ABE7-C5BF1A2C7675@voicestream.fi>
Message-ID: <96A560F3BDAA4BDF90A7BD3B0E3297CC@dell9400>

Hi Ari,

Firstly - I have actually been involved with 2 E1/T1 card vendors earlier,
so I am happy to make a few suggestions.

But, firstly - 

1. What DSP and RTOS are you using? 
2. What is the expected price range of the cards?
3. When will the cards be available?
4. Any chance you guys will provide open source on firmware/driver so we can
develop on it and get it right :)?
5. Any block schema you can share? I am happy to sign a NDA.
6. Is this PCI only or are you doing a stand-alone card as well?

Nice to see that someone actually target the open source marked with DSP
based cards.

Myself, I really would like an DSP based, open source project. 

Jan

-----Original Message-----
From: freeswitch-dev-bounces at lists.freeswitch.org
[mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Ari
Siitonen
Sent: 16. august 2010 09:07
To: freeswitch-dev at lists.freeswitch.org
Subject: [Freeswitch-dev] New Hardware for Freeswitch

Hello all!

I would like to announce a new hardware development project:

- ISDN Pri interface card 1-8 channels with/without QUICC for ISDN  
stack + dsp
- 1-8 POTS line card
- ISDN Bri line card network/user side

Our company has manufactured such cards and IVR-server back in last  
century,
and we would like to make the "new generation" cards compatible with
freeswitch.

So, to start Hardware redevelopment in right direction, any pointers and
suggestions on driver design would be greatly appreciated. And if the
existing cards lack some features, we will try to incorporate them (hw
conference bridging?)

Also, pilot users will be given free cards to test and evaluate.



Best Regards,

Ari Siitonen
Voice Stream Oy
Finland

_______________________________________________
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 aroumie at yahoo.com  Mon Aug 16 14:44:22 2010
From: aroumie at yahoo.com (Ali R.)
Date: Mon, 16 Aug 2010 14:44:22 -0700 (PDT)
Subject: [Freeswitch-dev] Socket - bridge answered event
In-Reply-To: <AANLkTikH4dsW9Ym8sCe_iqOQEdBBTjj40GJPvXohmg2A@mail.gmail.com>
References: <250201.55875.qm@web120608.mail.ne1.yahoo.com>
	<AANLkTimxj-oRveqM9CtYdLSs2ynEbeQ0aqOrxY0z=n_u@mail.gmail.com>
	<483598.28088.qm@web120619.mail.ne1.yahoo.com>
	<AANLkTikH4dsW9Ym8sCe_iqOQEdBBTjj40GJPvXohmg2A@mail.gmail.com>
Message-ID: <370568.21254.qm@web120611.mail.ne1.yahoo.com>

Hello,
I'm looking for this event (LEG B CHANNEL_ANSWER) just to update a UI.? Right 
now, I'm shifting my logic to use the inbound mode where I see all events 
including when Leg B is answered.
Thanks Again




________________________________
From: lakshmanan ganapathy <lakindia89 at gmail.com>
To: freeswitch-dev at lists.freeswitch.org
Sent: Thu, August 12, 2010 12:44:03 AM
Subject: Re: [Freeswitch-dev] Socket - bridge answered event

Ok. I understand your point.

One more way is, when the LEG B answers the call, you will get CHANNEL_BRIDGE 
event in the A LEG. There you will have all the informations regarding the LEG B 
( including the UUID of the LEG B ).

May I know What you want to do when LEB B is answered??, so that I can suggest 
better ways.




On Thu, Aug 12, 2010 at 1:00 PM, Ali R. <aroumie at yahoo.com> wrote:

Thank you so much for your response!
>I have not tried what you suggested but don?t you think sending the command
>"events plain all" on an outbound socket creates lots of?overhead on my app
>and the FS event socket module??
>Correct me if I'm wrong, in outbound socket mode FS spawns a new socket
>connection for each Leg A into my listening server?? Also?if I pass
>"events plain all" or the other handy command "events xml all" on each
>socket, FS will push the same TCP stream over?all connected sockets.?
>For example, if I got 50 connected Leg A, FS will push the same event?50
>times?to my listening?socket?? My app is very sensitive to bandwidth and
>that's?the reason I set the filter to just filter out the events that are
>sufficient
>for my application logic.
>I'm still experiencing with FS and I might be wrong.
>
>P.S: Regarding my issue, I noticed when Leg B is answered the fs_cli logs this
>event and that's the one I'm looking for?at my?end point socket.
>Thanks,
>Ali,
>
>
>
>________________________________
>From: lakshmanan ganapathy <lakindia89 at gmail.com>
>To: freeswitch-dev at lists.freeswitch.org
>Sent: Wed, August 11, 2010 9:24:32 PM
>Subject: Re: [Freeswitch-dev] Socket - bridge answered event
>
>
>Hi
>
>You register for myevents only.
>So you will receive events only for LEG A.
>Just try
>??? events plain all
>
>
>
>
>
>On Wed, Aug 11, 2010 at 8:00 AM, Ali Roumie <aroumie at yahoo.com> wrote:
>
>Hello All,
>>This is my first post to this list and many thanks to all contributors to this
>>state of art project. I'm using outbound socket and everything is going
>>wonderful with me except one thing so far.
>>My logic is simple, collect a PIN from Leg A and once the PIN is authorized, I
>>bridge the call with a SIP provider. Everything is great, call got bridged
>>successfully (and I was supper exited when it worked) but my problems is FS is
>>not sending my socket an event when the Leg B is answered.? I get an event 
only
>>when Leg B is hangup.? I must mention, I set filters on the socket to avoid
>lots
>>of the many generated events by FS
>>Here is my command list on the socket.
>>?
>>connect
>>myevents
>>filter Event-Name CHANNEL_ANSWER
>>filter Event-Name CHANNEL_CREATE
>>filter Event-Name CHANNEL_EXECUTE_COMPLETE
>>filter Event-Name CHANNEL_BRIDGE
>>filter Event-Name CHANNEL_UNBRIDGE
>>filter Event-Name CHANNEL_HANGUP
>>sendmsg
>>call-command: execute
>>execute-app-name: answer
>>sendmsg
>>call-command: execute
>>execute-app-name: sleep
>>execute-app-arg:1000
>>event-lock:true
>>sendmsg
>>call-command: execute
>>execute-app-name: play_and_get_digits
>>execute-app-arg:2 5 3 5000 # ivr/ivr-please_enter_pin_followed_by_pound.wav
>>ivr/ivr-please_reenter_your_pin.wav chanDTMF ^\d{5}$
>>event-lock:true
>>sendmsg
>>call-command: execute
>>execute-app-name: bridge
>>execute-app-arg:[sip_auth_username=XXXX,sip_auth_password=*******]sofia/internal/123456789 at PROVIDER.COM
>>
>>
>>
>>
>>
>>event-lock:true
>>?
>>Many Thanks,
>>Ali R.
>>
>>
>>
>>
>>_______________________________________________
>>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
>>
>
>
>
>
>
>
>_______________________________________________
>FreeSWITCH-dev mailing list
>FreeSWITCH-dev at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>http://www.freeswitch.org
>



      
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From msc at freeswitch.org  Wed Aug 18 09:42:17 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 18 Aug 2010 09:42:17 -0700
Subject: [Freeswitch-dev] FreeSWITCH Conference Call Starting Shortly!
Message-ID: <AANLkTinB0+q322Z3CnHFyDW=vYFDTr-rby8413LXPZR=@mail.gmail.com>

C'mon down!

http://wiki.freeswitch.org/wiki/FS_weekly_2010_08_18

Brian K West will be by to talk about NAT and VoIP with FreeSWITCH.

-MC
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From stamm at lyth.de  Tue Aug 24 10:54:11 2010
From: stamm at lyth.de (Achim Stamm)
Date: Tue, 24 Aug 2010 19:54:11 +0200
Subject: [Freeswitch-dev] Freeswitch communication delay after three hours
Message-ID: <4C740743.3000901@lyth.de>

Hello!

I have following problem:

My current session is connected to a thirdParty Application by using an 
own FreeSwitch
application "DoingReadAndWriteFrames". The communication between 
FreeSwitch Session and
thirdParty Application is done by reading and writing frames in a while 
loop.
At start of FreeSwitch Session the communication works without delay.
After three or more hours i get a communication delay of one or more 
seconds.
If i hang up and make a new call (a new freeswitch session), than the 
delay disappears, but after several hours
the delay comes up again.

It is possible, that i get an old frame (one second ago) with 
switch_core_session_read_frame ?

Is there a solution (for example an freeswitch core api call) to avoid 
the delay ?

Greetings

Achim Stamm

-- 
Achim Stamm, Dipl.-Inform. (FH)


Lyncker & Theis GmbH
Wilhelmstr. 16
65185 Wiesbaden
Germany

Fon +49 611/9006951
Fax +49 611/9406125


Handelsregister: HRB 23156 Amtsgericht Wiesbaden
Steuernummer: 04323897052
USt-IdNr.: DE255806399

Gesch?ftsf?hrer:
Filip Lyncker,
Armin Theis 



From anthony.minessale at gmail.com  Tue Aug 24 11:07:04 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 24 Aug 2010 13:07:04 -0500
Subject: [Freeswitch-dev] Freeswitch communication delay after three
	hours
In-Reply-To: <4C740743.3000901@lyth.de>
References: <4C740743.3000901@lyth.de>
Message-ID: <AANLkTin=YvF-QohsAqKtkiA6EL2rv7XswN9H8cOn5Rf2@mail.gmail.com>

Is this over SIP?


On Tue, Aug 24, 2010 at 12:54 PM, Achim Stamm <stamm at lyth.de> wrote:
> Hello!
>
> I have following problem:
>
> My current session is connected to a thirdParty Application by using an
> own FreeSwitch
> application "DoingReadAndWriteFrames". The communication between
> FreeSwitch Session and
> thirdParty Application is done by reading and writing frames in a while
> loop.
> At start of FreeSwitch Session the communication works without delay.
> After three or more hours i get a communication delay of one or more
> seconds.
> If i hang up and make a new call (a new freeswitch session), than the
> delay disappears, but after several hours
> the delay comes up again.
>
> It is possible, that i get an old frame (one second ago) with
> switch_core_session_read_frame ?
>
> Is there a solution (for example an freeswitch core api call) to avoid
> the delay ?
>
> Greetings
>
> Achim Stamm
>
> --
> Achim Stamm, Dipl.-Inform. (FH)
>
>
> Lyncker & Theis GmbH
> Wilhelmstr. 16
> 65185 Wiesbaden
> Germany
>
> Fon +49 611/9006951
> Fax +49 611/9406125
>
>
> Handelsregister: HRB 23156 Amtsgericht Wiesbaden
> Steuernummer: 04323897052
> USt-IdNr.: DE255806399
>
> Gesch?ftsf?hrer:
> Filip Lyncker,
> Armin Theis
>
>
> _______________________________________________
> 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 mrene_lists at avgs.ca  Tue Aug 24 15:19:38 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Tue, 24 Aug 2010 18:19:38 -0400
Subject: [Freeswitch-dev] Freeswitch communication delay after three
	hours
In-Reply-To: <4C740743.3000901@lyth.de>
References: <4C740743.3000901@lyth.de>
Message-ID: <012A802C-3F37-4B6F-8F3D-7BD87CFB0A50@avgs.ca>

Hi,

Lange nicht gesprochen. What can happen, usually, is network congestion causing a delay buildup over time (FS times its reads at every 20ms and returns a comfort noise frame if it can't get an RTP frame immediately). If frames come in 2-3 ms delayed because of congestion, delay can build up over time. If you are in a C module, you can send a message to the channel so it drops frames (essentially reads until EWOULDBLOCK).

zB:

switch_core_session_message_t msg = { 0 };
msg.message_id = SWITCH_MESSAGE_INDICATE_AUDIO_SYNC;
msg.from = __FILE__;
switch_core_session_receive_message(session, &msg); 

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




On 2010-08-24, at 1:54 PM, Achim Stamm wrote:

> Hello!
> 
> I have following problem:
> 
> My current session is connected to a thirdParty Application by using an 
> own FreeSwitch
> application "DoingReadAndWriteFrames". The communication between 
> FreeSwitch Session and
> thirdParty Application is done by reading and writing frames in a while 
> loop.
> At start of FreeSwitch Session the communication works without delay.
> After three or more hours i get a communication delay of one or more 
> seconds.
> If i hang up and make a new call (a new freeswitch session), than the 
> delay disappears, but after several hours
> the delay comes up again.
> 
> It is possible, that i get an old frame (one second ago) with 
> switch_core_session_read_frame ?
> 
> Is there a solution (for example an freeswitch core api call) to avoid 
> the delay ?
> 
> Greetings
> 
> Achim Stamm
> 
> -- 
> Achim Stamm, Dipl.-Inform. (FH)
> 
> 
> Lyncker & Theis GmbH
> Wilhelmstr. 16
> 65185 Wiesbaden
> Germany
> 
> Fon +49 611/9006951
> Fax +49 611/9406125
> 
> 
> Handelsregister: HRB 23156 Amtsgericht Wiesbaden
> Steuernummer: 04323897052
> USt-IdNr.: DE255806399
> 
> Gesch?ftsf?hrer:
> Filip Lyncker,
> Armin Theis 
> 
> 
> _______________________________________________
> 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 rjcajax at gmail.com  Wed Aug 25 05:39:21 2010
From: rjcajax at gmail.com (Robert Clayton)
Date: Wed, 25 Aug 2010 08:39:21 -0400
Subject: [Freeswitch-dev] Unsubscribe
Message-ID: <AANLkTikOdA2rUc-nEOFpaLJARyP-Sv+dbw4+xMA0iFrs@mail.gmail.com>

When moving both my Freeswitch dev and user accounts it appears that
one must log in to unsubscribe as both the unsubscribe and remind
(password) buttons do not work. The problem being if a person does not
remember their password they would not be able to unsubscribe. Which
is the problem with my dev account.


From sushil_kv2004 at yahoo.com  Tue Aug 24 10:20:39 2010
From: sushil_kv2004 at yahoo.com (sushil verma)
Date: Tue, 24 Aug 2010 10:20:39 -0700 (PDT)
Subject: [Freeswitch-dev] How is MOH implemented in Freeswitch
Message-ID: <671124.24379.qm@web111506.mail.gq1.yahoo.com>


??? Hi,
??????? It seems Freeswitch implements MOH (music on hold) feature using sofia module.

??????? Reference: http://wiki.freeswitch.org/wiki/Music_on_Hold

????? Can someone explain (or point me to some design/feature document etc of freeswitch) how it has been implemented?

??? Regards,
??? Sushil Kumar







      
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From brian at freeswitch.org  Wed Aug 25 10:29:13 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 25 Aug 2010 12:29:13 -0500
Subject: [Freeswitch-dev] Unsubscribe
In-Reply-To: <AANLkTikOdA2rUc-nEOFpaLJARyP-Sv+dbw4+xMA0iFrs@mail.gmail.com>
References: <AANLkTikOdA2rUc-nEOFpaLJARyP-Sv+dbw4+xMA0iFrs@mail.gmail.com>
Message-ID: <38126DF6-6D72-4949-9E16-D3ED8763F102@freeswitch.org>

The password isn't required.. you can do it via email too and it will email you a link to confirm.

/b

On Aug 25, 2010, at 7:39 AM, Robert Clayton wrote:

> When moving both my Freeswitch dev and user accounts it appears that
> one must log in to unsubscribe as both the unsubscribe and remind
> (password) buttons do not work. The problem being if a person does not
> remember their password they would not be able to unsubscribe. Which
> is the problem with my dev account.



From gmaruzz at celliax.org  Wed Aug 25 10:33:04 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 25 Aug 2010 19:33:04 +0200
Subject: [Freeswitch-dev] Unsubscribe
In-Reply-To: <AANLkTikOdA2rUc-nEOFpaLJARyP-Sv+dbw4+xMA0iFrs@mail.gmail.com>
References: <AANLkTikOdA2rUc-nEOFpaLJARyP-Sv+dbw4+xMA0iFrs@mail.gmail.com>
Message-ID: <AANLkTimOQ=Dw3HbFys_q=8f30CynymtfR5EoeHnbZ+cX@mail.gmail.com>

On Wed, Aug 25, 2010 at 2:39 PM, Robert Clayton <rjcajax at gmail.com> wrote:
> When moving both my Freeswitch dev and user accounts it appears that
> one must log in to unsubscribe as both the unsubscribe and remind
> (password) buttons do not work. The problem being if a person does not
> remember their password they would not be able to unsubscribe. Which
> is the problem with my dev account.

Is awful when you realize you're growing old, eh?

:P

-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From jkr888 at gmail.com  Wed Aug 25 13:46:54 2010
From: jkr888 at gmail.com (Johny Kadarisman Kwan)
Date: Wed, 25 Aug 2010 16:46:54 -0400
Subject: [Freeswitch-dev] FS complained on RTP packet.
Message-ID: <AANLkTimztouRpHws8s8eOrWDkemUSBVrGG4kLpEEqqc+@mail.gmail.com>

I'm trying to convert a proprietary audio stream into sip/rtp compatible. At
this point, i'm able to pass sip negotiation with FS, and trying to stream
PCMU/8000 codec. But something not right on my end, and FS complained about
the ptime settings.
I'm sending 320 bytes ulaw payload on every rtp packet. Is that the right
amount? appreciate if anyone could point some info on these topics.

Thanks,
JK

===============
*2010-08-25 16:36:58.142328 [WARNING] mod_sofia.c:1013 We were told to use
ptime 20 but what they meant to say was 40*
*This issue has so far been identified to happen on the following broken
platforms/devices:*
*Linksys/Sipura aka Cisco*
*ShoreTel*
*Sonus/L3*
*We will try to fix it but some of the devices on this list are so broken,*
*who knows what will happen..*
*===================*
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From mike at jerris.com  Wed Aug 25 20:30:17 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 25 Aug 2010 23:30:17 -0400
Subject: [Freeswitch-dev] FS complained on RTP packet.
In-Reply-To: <AANLkTimztouRpHws8s8eOrWDkemUSBVrGG4kLpEEqqc+@mail.gmail.com>
References: <AANLkTimztouRpHws8s8eOrWDkemUSBVrGG4kLpEEqqc+@mail.gmail.com>
Message-ID: <4EBBAC6B-3476-4D36-81AA-96B6088B9614@jerris.com>

20 ms of ulaw would be 160 bytes, not 320.

Mike

On Aug 25, 2010, at 4:46 PM, Johny Kadarisman Kwan wrote:

> I'm trying to convert a proprietary audio stream into sip/rtp compatible. At this point, i'm able to pass sip negotiation with FS, and trying to stream PCMU/8000 codec. But something not right on my end, and FS complained about the ptime settings.
> I'm sending 320 bytes ulaw payload on every rtp packet. Is that the right amount? appreciate if anyone could point some info on these topics.
> 
> Thanks,
> JK
> 
> ===============
> 2010-08-25 16:36:58.142328 [WARNING] mod_sofia.c:1013 We were told to use ptime 20 but what they meant to say was 40
> This issue has so far been identified to happen on the following broken platforms/devices:
> Linksys/Sipura aka Cisco
> ShoreTel
> Sonus/L3
> We will try to fix it but some of the devices on this list are so broken,
> who knows what will happen..
> ===================

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From farhan.husain at csebuet.org  Wed Aug 25 22:27:33 2010
From: farhan.husain at csebuet.org (Farhan Husain)
Date: Thu, 26 Aug 2010 00:27:33 -0500
Subject: [Freeswitch-dev] Module written in C++
Message-ID: <AANLkTinY_3rzkZfLeJOZ5WwPVyG4Ha9kciWYmXdfHTQA@mail.gmail.com>

I am trying to write a module having multiple C++ source files. When I run
make, I find that only the C++ file having the name same as the module name
is being compiled and all others do not produce any object file. As a result
when I run FS I get linker error. Can anyone tell me how to get them
compiled too?

Thanks,
Farhan
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From peter.olsson at visionutveckling.se  Wed Aug 25 22:36:02 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 26 Aug 2010 07:36:02 +0200
Subject: [Freeswitch-dev] Module written in C++
In-Reply-To: <AANLkTinY_3rzkZfLeJOZ5WwPVyG4Ha9kciWYmXdfHTQA@mail.gmail.com>
References: <AANLkTinY_3rzkZfLeJOZ5WwPVyG4Ha9kciWYmXdfHTQA@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C57DC05816A@cooper>

Have you updated the Makefile to build the other files?

/Peter
________________________________________
Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [freeswitch-dev-bounces at lists.freeswitch.org] f&#246;r Farhan Husain [farhan.husain at csebuet.org]
Skickat: den 26 augusti 2010 07:27
Till: FreeSWITCH Users Help; freeswitch-dev at lists.freeswitch.org
?mne: [Freeswitch-dev] Module written in C++

I am trying to write a module having multiple C++ source files. When I run make, I find that only the C++ file having the name same as the module name is being compiled and all others do not produce any object file. As a result when I run FS I get linker error. Can anyone tell me how to get them compiled too?

Thanks,
Farhan
!DSPAM:4c75fc8232934761421429!


From farhan.husain at csebuet.org  Wed Aug 25 22:58:47 2010
From: farhan.husain at csebuet.org (Farhan Husain)
Date: Thu, 26 Aug 2010 00:58:47 -0500
Subject: [Freeswitch-dev] Module written in C++
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C57DC05816A@cooper>
References: <AANLkTinY_3rzkZfLeJOZ5WwPVyG4Ha9kciWYmXdfHTQA@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C57DC05816A@cooper>
Message-ID: <AANLkTikgy7A7AP4saHiCZtwbzHYuzjCc+aMHB=8mquR6@mail.gmail.com>

I just got the solution from Math at the IRC channel. Here is the link to
the solution he gave me: http://pastebin.freeswitch.org/13727

On Thu, Aug 26, 2010 at 12:36 AM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> Have you updated the Makefile to build the other files?
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [
> freeswitch-dev-bounces at lists.freeswitch.org] f&#246;r Farhan Husain [
> farhan.husain at csebuet.org]
> Skickat: den 26 augusti 2010 07:27
> Till: FreeSWITCH Users Help; freeswitch-dev at lists.freeswitch.org
> ?mne: [Freeswitch-dev] Module written in C++
>
> I am trying to write a module having multiple C++ source files. When I run
> make, I find that only the C++ file having the name same as the module name
> is being compiled and all others do not produce any object file. As a result
> when I run FS I get linker error. Can anyone tell me how to get them
> compiled too?
>
> Thanks,
> Farhan
> !DSPAM:4c75fc8232934761421429!
>
> _______________________________________________
> 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 steveayre at gmail.com  Wed Aug 25 23:23:06 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 26 Aug 2010 07:23:06 +0100
Subject: [Freeswitch-dev] Module written in C++
In-Reply-To: <AANLkTikgy7A7AP4saHiCZtwbzHYuzjCc+aMHB=8mquR6@mail.gmail.com>
References: <AANLkTinY_3rzkZfLeJOZ5WwPVyG4Ha9kciWYmXdfHTQA@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C57DC05816A@cooper>
	<AANLkTikgy7A7AP4saHiCZtwbzHYuzjCc+aMHB=8mquR6@mail.gmail.com>
Message-ID: <AANLkTinE8WGxUDOS4mfzt5GxfEzYpLdc2zX=mCGxZ9Rg@mail.gmail.com>

I'll just copy and paste that for the archives / my own records...

   1. BASE=../../../..
   2. LOCAL_CFLAGS=-Wall -Werror
   3. LOCAL_LDFLAGS=
   4. LOCAL_OBJS=file.o file2.o file3.o
   5.
   6. local_depend: $(LOCAL_OBJS)
   7.
   8. include $(BASE)/build/modmake.rules


-Steve

On 26 August 2010 06:58, Farhan Husain <farhan.husain at csebuet.org> wrote:

> I just got the solution from Math at the IRC channel. Here is the link to
> the solution he gave me: http://pastebin.freeswitch.org/13727
>
>
> On Thu, Aug 26, 2010 at 12:36 AM, Peter Olsson <
> peter.olsson at visionutveckling.se> wrote:
>
>> Have you updated the Makefile to build the other files?
>>
>> /Peter
>> ________________________________________
>> Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [
>> freeswitch-dev-bounces at lists.freeswitch.org] f&#246;r Farhan Husain [
>> farhan.husain at csebuet.org]
>> Skickat: den 26 augusti 2010 07:27
>> Till: FreeSWITCH Users Help; freeswitch-dev at lists.freeswitch.org
>> ?mne: [Freeswitch-dev] Module written in C++
>>
>> I am trying to write a module having multiple C++ source files. When I run
>> make, I find that only the C++ file having the name same as the module name
>> is being compiled and all others do not produce any object file. As a result
>> when I run FS I get linker error. Can anyone tell me how to get them
>> compiled too?
>>
>> Thanks,
>> Farhan
>> !DSPAM:4c75fc8232934761421429!
>>
>> _______________________________________________
>> 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
>>
>
>
> _______________________________________________
> 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 steveayre at gmail.com  Thu Aug 26 01:13:08 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 26 Aug 2010 09:13:08 +0100
Subject: [Freeswitch-dev] FS complained on RTP packet.
In-Reply-To: <4EBBAC6B-3476-4D36-81AA-96B6088B9614@jerris.com>
References: <AANLkTimztouRpHws8s8eOrWDkemUSBVrGG4kLpEEqqc+@mail.gmail.com>
	<4EBBAC6B-3476-4D36-81AA-96B6088B9614@jerris.com>
Message-ID: <AANLkTikUan6sxDhwAyrUUpdRrrt9WAh=3dvgzYGxQ5Dn@mail.gmail.com>

Which is 40ms... which is what FS tells you. FS will look at the amount
*actually* received, instead of the amount the client claimed it will sent.
It auto-adjusts to that amount, and displays the warning message because
your client is either misconfigured or broken (some SIP phone
implementations lie).

Because it auto-adjusts it shouldn't be an issue, but it could also affect
quality so a warning is given just in case so that you can fix the problem
on the client.

-Steve


On 26 August 2010 04:30, Michael Jerris <mike at jerris.com> wrote:

> 20 ms of ulaw would be 160 bytes, not 320.
>
> Mike
>
> On Aug 25, 2010, at 4:46 PM, Johny Kadarisman Kwan wrote:
>
> I'm trying to convert a proprietary audio stream into sip/rtp compatible.
> At this point, i'm able to pass sip negotiation with FS, and trying to
> stream PCMU/8000 codec. But something not right on my end, and FS complained
> about the ptime settings.
> I'm sending 320 bytes ulaw payload on every rtp packet. Is that the right
> amount? appreciate if anyone could point some info on these topics.
>
> Thanks,
> JK
>
> ===============
> *2010-08-25 16:36:58.142328 [WARNING] mod_sofia.c:1013 We were told to use
> ptime 20 but what they meant to say was 40*
> *This issue has so far been identified to happen on the following broken
> platforms/devices:*
> *Linksys/Sipura aka Cisco*
> *ShoreTel*
> *Sonus/L3*
> *We will try to fix it but some of the devices on this list are so broken,
> *
> *who knows what will happen..*
> *===================*
>
>
>
> _______________________________________________
> 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 stamm at lyth.de  Thu Aug 26 04:19:44 2010
From: stamm at lyth.de (Achim Stamm)
Date: Thu, 26 Aug 2010 13:19:44 +0200
Subject: [Freeswitch-dev] Freeswitch communication delay after
	three	hours
In-Reply-To: <012A802C-3F37-4B6F-8F3D-7BD87CFB0A50@avgs.ca>
References: <4C740743.3000901@lyth.de>
	<012A802C-3F37-4B6F-8F3D-7BD87CFB0A50@avgs.ca>
Message-ID: <4C764DD0.4080701@lyth.de>

Mathieu Rene schrieb:
> Hi,
>
> Lange nicht gesprochen. What can happen, usually, is network congestion causing a delay buildup over time (FS times its reads at every 20ms and returns a comfort noise frame if it can't get an RTP frame immediately). If frames come in 2-3 ms delayed because of congestion, delay can build up over time. If you are in a C module, you can send a message to the channel so it drops frames (essentially reads until EWOULDBLOCK).
>
> zB:
>
> switch_core_session_message_t msg = { 0 };
> msg.message_id = SWITCH_MESSAGE_INDICATE_AUDIO_SYNC;
> msg.from = __FILE__;
> switch_core_session_receive_message(session, &msg); 
>
> Hope that fixes it
>  
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 2010-08-24, at 1:54 PM, Achim Stamm wrote:
>
>   
>> Hello!
>>
>> I have following problem:
>>
>> My current session is connected to a thirdParty Application by using an 
>> own FreeSwitch
>> application "DoingReadAndWriteFrames". The communication between 
>> FreeSwitch Session and
>> thirdParty Application is done by reading and writing frames in a while 
>> loop.
>> At start of FreeSwitch Session the communication works without delay.
>> After three or more hours i get a communication delay of one or more 
>> seconds.
>> If i hang up and make a new call (a new freeswitch session), than the 
>> delay disappears, but after several hours
>> the delay comes up again.
>>
>> It is possible, that i get an old frame (one second ago) with 
>> switch_core_session_read_frame ?
>>
>> Is there a solution (for example an freeswitch core api call) to avoid 
>> the delay ?
>>
>> Greetings
>>
>> Achim Stamm
>>
>> -- 
>> Achim Stamm, Dipl.-Inform. (FH)
>>
>>
>> Lyncker & Theis GmbH
>> Wilhelmstr. 16
>> 65185 Wiesbaden
>> Germany
>>
>> Fon +49 611/9006951
>> Fax +49 611/9406125
>>
>>
>> Handelsregister: HRB 23156 Amtsgericht Wiesbaden
>> Steuernummer: 04323897052
>> USt-IdNr.: DE255806399
>>
>> Gesch?ftsf?hrer:
>> Filip Lyncker,
>> Armin Theis 
>>
>>
>> _______________________________________________
>> 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
>>     
>
>
> _______________________________________________
> 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
>
>   
Thanks for your suggestion.

The fix works, but when should i do the fix ?

Best way for me is to call "Audio Sync" every hour (or every ten 
minutes) after session is started and when silence is on channel.

First try was to place audio synchronisation, when i get a silent Frame:
if (switch_test_flag(read_frame, SFF_CNG) || !read_frame->samples)
{
    switch_core_session_message_t msg = { 0 };
    msg.message_id = SWITCH_MESSAGE_INDICATE_AUDIO_SYNC;
    msg.from = __FILE__;
    switch_core_session_receive_message(session, &msg);
}
Here the problem is, that i don't get an silence Frame, so audio 
synchronisation is never called.

How can i recognize silent on channel ?

Next try was to call every 10 minutes to do the "Audio Sync".
But if somebody says something at same time doing audio synchronisation, 
this frames will be droped.

How can i get the session running time ?


Greetings

Achim Stamm


-- 
Achim Stamm, Dipl.-Inform. (FH)


Lyncker & Theis GmbH
Wilhelmstr. 16
65185 Wiesbaden
Germany

Fon +49 611/9006951
Fax +49 611/9406125


Handelsregister: HRB 23156 Amtsgericht Wiesbaden
Steuernummer: 04323897052
USt-IdNr.: DE255806399

Gesch?ftsf?hrer:
Filip Lyncker,
Armin Theis 



From trixter at 0xdecafbad.com  Thu Aug 26 05:11:46 2010
From: trixter at 0xdecafbad.com (Bret McDanel)
Date: Thu, 26 Aug 2010 05:11:46 -0700
Subject: [Freeswitch-dev] Freeswitch communication delay after three
 hours
In-Reply-To: <4C764DD0.4080701@lyth.de>
References: <4C740743.3000901@lyth.de>
	<012A802C-3F37-4B6F-8F3D-7BD87CFB0A50@avgs.ca>
	<4C764DD0.4080701@lyth.de>
Message-ID: <1282824706.2479.5.camel@trixeee.0xdecafbad.com>

On Thu, 2010-08-26 at 13:19 +0200, Achim Stamm wrote:
> Mathieu Rene schrieb:
> > Hi,
> >
> > Lange nicht gesprochen. What can happen, usually, is network
> congestion causing a delay buildup over time (FS times its reads at
> every 20ms and returns a comfort noise frame if it can't get an RTP
> frame immediately). If frames come in 2-3 ms delayed because of
> congestion, delay can build up over time. If you are in a C module,
> you can send a message to the channel so it drops frames (essentially
> reads until EWOULDBLOCK).
> >


I am unsure if your explanation is correct.  If there is a delay of 2-3
ms 1 comfort noise frame should be generated (unless its doing
discontinuous media), after that it should have a buffer built up of
received frames so another delay of 2-3 ms would not cause an
interleaving of CNG and audio (which would degrade call quality).


-- 
Trixter aka Bret McDanel
website:  http://www.0xdecafbad.com
pgp key:  http://bit.ly/9XYK4b



From dave at 3c.co.uk  Thu Aug 26 05:59:09 2010
From: dave at 3c.co.uk (David Knell)
Date: Thu, 26 Aug 2010 15:59:09 +0300
Subject: [Freeswitch-dev] Freeswitch communication delay after three
 hours
In-Reply-To: <1282824706.2479.5.camel@trixeee.0xdecafbad.com>
References: <4C740743.3000901@lyth.de>
	<012A802C-3F37-4B6F-8F3D-7BD87CFB0A50@avgs.ca>
	<4C764DD0.4080701@lyth.de>
	<1282824706.2479.5.camel@trixeee.0xdecafbad.com>
Message-ID: <1282827549.2679.7.camel@dk-d820>

On Thu, 2010-08-26 at 05:11 -0700, Bret McDanel wrote:
> On Thu, 2010-08-26 at 13:19 +0200, Achim Stamm wrote:
> > Mathieu Rene schrieb:
> > > Hi,
> > >
> > > Lange nicht gesprochen. What can happen, usually, is network
> > congestion causing a delay buildup over time (FS times its reads at
> > every 20ms and returns a comfort noise frame if it can't get an RTP
> > frame immediately). If frames come in 2-3 ms delayed because of
> > congestion, delay can build up over time. If you are in a C module,
> > you can send a message to the channel so it drops frames (essentially
> > reads until EWOULDBLOCK).
> > >
> 
> 
> I am unsure if your explanation is correct.  If there is a delay of 2-3
> ms 1 comfort noise frame should be generated (unless its doing
> discontinuous media), after that it should have a buffer built up of
> received frames so another delay of 2-3 ms would not cause an
> interleaving of CNG and audio (which would degrade call quality).

Have to agree with Bret here - the thing which causes noticeable delays
to start to appear on long calls is a mismatch in clocks between two
points on the chain.  One end sending packets slightly faster than the
other processes them will result in an indefinite buildup in delay over
time, unless:
- the jitter buffering's clever enough to notice this, and chuck packets
away;
- the sender has silence suppression on (as in send nothing, not send
CNG frames), and the party at that end occasionally stops talking (cue
mother-in-law joke) at which point things get back in sync;
- (possibly) there's a non-zero amount of packet loss, and the receiver
doesn't try to fill the gaps by duplicating previous packets.

--Dave

-- 
David Knell, Director, 3C Limited
T: +44 20 3298 2000
E: dave at 3c.co.uk
W: http://www.3c.co.uk



From jkr888 at gmail.com  Thu Aug 26 06:54:04 2010
From: jkr888 at gmail.com (Johny Kadarisman Kwan)
Date: Thu, 26 Aug 2010 09:54:04 -0400
Subject: [Freeswitch-dev] FS complained on RTP packet.
In-Reply-To: <AANLkTikUan6sxDhwAyrUUpdRrrt9WAh=3dvgzYGxQ5Dn@mail.gmail.com>
References: <AANLkTimztouRpHws8s8eOrWDkemUSBVrGG4kLpEEqqc+@mail.gmail.com>
	<4EBBAC6B-3476-4D36-81AA-96B6088B9614@jerris.com>
	<AANLkTikUan6sxDhwAyrUUpdRrrt9WAh=3dvgzYGxQ5Dn@mail.gmail.com>
Message-ID: <AANLkTi=b3mB-SyhCEojmqMg7vVS1ULKL1-3cNQaVuZHB@mail.gmail.com>

Its excellent that  FS able to auto adjust to 40ms.Sending 160 bytes payload
does eliminate warning message from FS.
My audio doesn't work yet, problem must be something else. At least no more
issues on rtp audio framing ;)

Thanks,
JK


On Thu, Aug 26, 2010 at 4:13 AM, Steven Ayre <steveayre at gmail.com> wrote:

> Which is 40ms... which is what FS tells you. FS will look at the amount
> *actually* received, instead of the amount the client claimed it will sent.
> It auto-adjusts to that amount, and displays the warning message because
> your client is either misconfigured or broken (some SIP phone
> implementations lie).
>
> Because it auto-adjusts it shouldn't be an issue, but it could also affect
> quality so a warning is given just in case so that you can fix the problem
> on the client.
>
> -Steve
>
>
> On 26 August 2010 04:30, Michael Jerris <mike at jerris.com> wrote:
>
>> 20 ms of ulaw would be 160 bytes, not 320.
>>
>> Mike
>>
>> On Aug 25, 2010, at 4:46 PM, Johny Kadarisman Kwan wrote:
>>
>> I'm trying to convert a proprietary audio stream into sip/rtp compatible.
>> At this point, i'm able to pass sip negotiation with FS, and trying to
>> stream PCMU/8000 codec. But something not right on my end, and FS complained
>> about the ptime settings.
>> I'm sending 320 bytes ulaw payload on every rtp packet. Is that the right
>> amount? appreciate if anyone could point some info on these topics.
>>
>> Thanks,
>> JK
>>
>> ===============
>> *2010-08-25 16:36:58.142328 [WARNING] mod_sofia.c:1013 We were told to
>> use ptime 20 but what they meant to say was 40*
>> *This issue has so far been identified to happen on the following broken
>> platforms/devices:*
>> *Linksys/Sipura aka Cisco*
>> *ShoreTel*
>> *Sonus/L3*
>> *We will try to fix it but some of the devices on this list are so
>> broken,*
>> *who knows what will happen..*
>> *===================*
>>
>>
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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 brian at freeswitch.org  Thu Aug 26 07:02:05 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 26 Aug 2010 09:02:05 -0500
Subject: [Freeswitch-dev] FS complained on RTP packet.
In-Reply-To: <AANLkTi=b3mB-SyhCEojmqMg7vVS1ULKL1-3cNQaVuZHB@mail.gmail.com>
References: <AANLkTimztouRpHws8s8eOrWDkemUSBVrGG4kLpEEqqc+@mail.gmail.com>
	<4EBBAC6B-3476-4D36-81AA-96B6088B9614@jerris.com>
	<AANLkTikUan6sxDhwAyrUUpdRrrt9WAh=3dvgzYGxQ5Dn@mail.gmail.com>
	<AANLkTi=b3mB-SyhCEojmqMg7vVS1ULKL1-3cNQaVuZHB@mail.gmail.com>
Message-ID: <EEA61F66-395B-4A53-8BF5-AEA111898FF4@freeswitch.org>

Its could be your timestamps too... how many are you incrementing on each time stamp?  If you lie about time timestamps say send timestamps that jump by 320 but only send 160 byte payload you're still going to get the warning I'm pretty sure.

/b

On Aug 26, 2010, at 8:54 AM, Johny Kadarisman Kwan wrote:

> Its excellent that  FS able to auto adjust to 40ms.Sending 160 bytes payload does eliminate warning message from FS. 
> My audio doesn't work yet, problem must be something else. At least no more issues on rtp audio framing ;)
> 
> Thanks,
> JK



From jkr888 at gmail.com  Thu Aug 26 07:15:04 2010
From: jkr888 at gmail.com (Johny Kadarisman Kwan)
Date: Thu, 26 Aug 2010 10:15:04 -0400
Subject: [Freeswitch-dev] FS complained on RTP packet.
In-Reply-To: <EEA61F66-395B-4A53-8BF5-AEA111898FF4@freeswitch.org>
References: <AANLkTimztouRpHws8s8eOrWDkemUSBVrGG4kLpEEqqc+@mail.gmail.com>
	<4EBBAC6B-3476-4D36-81AA-96B6088B9614@jerris.com>
	<AANLkTikUan6sxDhwAyrUUpdRrrt9WAh=3dvgzYGxQ5Dn@mail.gmail.com>
	<AANLkTi=b3mB-SyhCEojmqMg7vVS1ULKL1-3cNQaVuZHB@mail.gmail.com>
	<EEA61F66-395B-4A53-8BF5-AEA111898FF4@freeswitch.org>
Message-ID: <AANLkTi=PbPg+v5dsG_pJxrv24gEdWH_mK=STfpMkNuWf@mail.gmail.com>

I adjust rtp timestamp to increment += 160, still no good audio.
i took code that handle speex previously, timestamp was set to 320increment.
seems working fine with speex/16k

still no good ulaw audio. I'm converting up stream audio that sent to me in
a large chunk, do some processing and now breaking up into smaller rtp
chunk. do i have to limit the rtp paket rate to freeswitch (ie. 20ms delay
between them)?

Thanks again.

On Thu, Aug 26, 2010 at 10:02 AM, Brian West <brian at freeswitch.org> wrote:

> Its could be your timestamps too... how many are you incrementing on each
> time stamp?  If you lie about time timestamps say send timestamps that jump
> by 320 but only send 160 byte payload you're still going to get the warning
> I'm pretty sure.
>
> /b
>
> On Aug 26, 2010, at 8:54 AM, Johny Kadarisman Kwan wrote:
>
> > Its excellent that  FS able to auto adjust to 40ms.Sending 160 bytes
> payload does eliminate warning message from FS.
> > My audio doesn't work yet, problem must be something else. At least no
> more issues on rtp audio framing ;)
> >
> > Thanks,
> > JK
>
>
> _______________________________________________
> 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 Aug 26 08:56:00 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 26 Aug 2010 08:56:00 -0700
Subject: [Freeswitch-dev] Everyone come join IRC!
Message-ID: <AANLkTin0Y+iBoxY+faQt3EL+1SNfoq1Ncy-T2cjo82rH@mail.gmail.com>

Hey all,

We are having a big IRC day today so we'd love to have everyone join and
help answer questions. Please join #freeswitch on irc.freenode.net.

Thanks!
-MC
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From anthony.minessale at gmail.com  Thu Aug 26 09:44:41 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 26 Aug 2010 11:44:41 -0500
Subject: [Freeswitch-dev] Freeswitch communication delay after three
	hours
In-Reply-To: <1282827549.2679.7.camel@dk-d820>
References: <4C740743.3000901@lyth.de>
	<012A802C-3F37-4B6F-8F3D-7BD87CFB0A50@avgs.ca>
	<4C764DD0.4080701@lyth.de>
	<1282824706.2479.5.camel@trixeee.0xdecafbad.com>
	<1282827549.2679.7.camel@dk-d820>
Message-ID: <AANLkTi=o3Fsn8Kw=vSNwhnVTOeyJXV8ix2UfXcJ2GGE5@mail.gmail.com>

you could set autoflush=true in the sofia profile so if it detects
prolonged packet build up it flushes the buffers.
VoIP is fun ain't it!

On Thu, Aug 26, 2010 at 7:59 AM, David Knell <dave at 3c.co.uk> wrote:
> On Thu, 2010-08-26 at 05:11 -0700, Bret McDanel wrote:
>> On Thu, 2010-08-26 at 13:19 +0200, Achim Stamm wrote:
>> > Mathieu Rene schrieb:
>> > > Hi,
>> > >
>> > > Lange nicht gesprochen. What can happen, usually, is network
>> > congestion causing a delay buildup over time (FS times its reads at
>> > every 20ms and returns a comfort noise frame if it can't get an RTP
>> > frame immediately). If frames come in 2-3 ms delayed because of
>> > congestion, delay can build up over time. If you are in a C module,
>> > you can send a message to the channel so it drops frames (essentially
>> > reads until EWOULDBLOCK).
>> > >
>>
>>
>> I am unsure if your explanation is correct. ?If there is a delay of 2-3
>> ms 1 comfort noise frame should be generated (unless its doing
>> discontinuous media), after that it should have a buffer built up of
>> received frames so another delay of 2-3 ms would not cause an
>> interleaving of CNG and audio (which would degrade call quality).
>
> Have to agree with Bret here - the thing which causes noticeable delays
> to start to appear on long calls is a mismatch in clocks between two
> points on the chain. ?One end sending packets slightly faster than the
> other processes them will result in an indefinite buildup in delay over
> time, unless:
> - the jitter buffering's clever enough to notice this, and chuck packets
> away;
> - the sender has silence suppression on (as in send nothing, not send
> CNG frames), and the party at that end occasionally stops talking (cue
> mother-in-law joke) at which point things get back in sync;
> - (possibly) there's a non-zero amount of packet loss, and the receiver
> doesn't try to fill the gaps by duplicating previous packets.
>
> --Dave
>
> --
> David Knell, Director, 3C Limited
> T: +44 20 3298 2000
> E: dave at 3c.co.uk
> W: http://www.3c.co.uk
>
>
> _______________________________________________
> 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 mike at jerris.com  Thu Aug 26 09:49:22 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 26 Aug 2010 12:49:22 -0400
Subject: [Freeswitch-dev] FS complained on RTP packet.
In-Reply-To: <AANLkTi=PbPg+v5dsG_pJxrv24gEdWH_mK=STfpMkNuWf@mail.gmail.com>
References: <AANLkTimztouRpHws8s8eOrWDkemUSBVrGG4kLpEEqqc+@mail.gmail.com>
	<4EBBAC6B-3476-4D36-81AA-96B6088B9614@jerris.com>
	<AANLkTikUan6sxDhwAyrUUpdRrrt9WAh=3dvgzYGxQ5Dn@mail.gmail.com>
	<AANLkTi=b3mB-SyhCEojmqMg7vVS1ULKL1-3cNQaVuZHB@mail.gmail.com>
	<EEA61F66-395B-4A53-8BF5-AEA111898FF4@freeswitch.org>
	<AANLkTi=PbPg+v5dsG_pJxrv24gEdWH_mK=STfpMkNuWf@mail.gmail.com>
Message-ID: <48B36522-3EB7-408F-BD65-ED22BE972D07@jerris.com>

If you are already getting them in larger chunks, you might as well pass along the packet size you get instead of breaking them up, just make sure to set the ptime correct.  What is the source of the audio?

Mike

On Aug 26, 2010, at 10:15 AM, Johny Kadarisman Kwan wrote:

> I adjust rtp timestamp to increment += 160, still no good audio. 
> i took code that handle speex previously, timestamp was set to 320increment. seems working fine with speex/16k
> 
> still no good ulaw audio. I'm converting up stream audio that sent to me in a large chunk, do some processing and now breaking up into smaller rtp chunk. do i have to limit the rtp paket rate to freeswitch (ie. 20ms delay between them)?
> 
> Thanks again.
> 
> On Thu, Aug 26, 2010 at 10:02 AM, Brian West <brian at freeswitch.org> wrote:
> Its could be your timestamps too... how many are you incrementing on each time stamp?  If you lie about time timestamps say send timestamps that jump by 320 but only send 160 byte payload you're still going to get the warning I'm pretty sure.
> 
> /b
> 
> On Aug 26, 2010, at 8:54 AM, Johny Kadarisman Kwan wrote:
> 
> > Its excellent that  FS able to auto adjust to 40ms.Sending 160 bytes payload does eliminate warning message from FS.
> > My audio doesn't work yet, problem must be something else. At least no more issues on rtp audio framing ;)
> >
> > Thanks,
> > JK
> 
> 
> _______________________________________________
> 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
> 
> _______________________________________________
> 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 jaybinks at gmail.com  Thu Aug 26 16:19:02 2010
From: jaybinks at gmail.com (jay binks)
Date: Fri, 27 Aug 2010 09:19:02 +1000
Subject: [Freeswitch-dev] Module written in C++
In-Reply-To: <AANLkTinE8WGxUDOS4mfzt5GxfEzYpLdc2zX=mCGxZ9Rg@mail.gmail.com>
References: <AANLkTinY_3rzkZfLeJOZ5WwPVyG4Ha9kciWYmXdfHTQA@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C57DC05816A@cooper>
	<AANLkTikgy7A7AP4saHiCZtwbzHYuzjCc+aMHB=8mquR6@mail.gmail.com>
	<AANLkTinE8WGxUDOS4mfzt5GxfEzYpLdc2zX=mCGxZ9Rg@mail.gmail.com>
Message-ID: <AANLkTikOiJGO+gsBFAVWUJtO=xhPufXNX2nhTrva3Utq@mail.gmail.com>

no... dont keep it for YOUR archives..

put it on the wiki..
( and post the url here please )

J


On Thu, Aug 26, 2010 at 4:23 PM, Steven Ayre <steveayre at gmail.com> wrote:
> I'll just copy and paste that for the archives / my own records...
>
> BASE=../../../..
> LOCAL_CFLAGS=-Wall -Werror
> LOCAL_LDFLAGS=
> LOCAL_OBJS=file.o file2.o file3.o
>
> local_depend: $(LOCAL_OBJS)
>
> include $(BASE)/build/modmake.rules
>
> -Steve
>
> On 26 August 2010 06:58, Farhan Husain <farhan.husain at csebuet.org> wrote:
>>
>> I just got the solution from Math at the IRC channel. Here is the link to
>> the solution he gave me:?http://pastebin.freeswitch.org/13727
>>
>> On Thu, Aug 26, 2010 at 12:36 AM, Peter Olsson
>> <peter.olsson at visionutveckling.se> wrote:
>>>
>>> Have you updated the Makefile to build the other files?
>>>
>>> /Peter
>>> ________________________________________
>>> Fr?n: freeswitch-dev-bounces at lists.freeswitch.org
>>> [freeswitch-dev-bounces at lists.freeswitch.org] f&#246;r Farhan Husain
>>> [farhan.husain at csebuet.org]
>>> Skickat: den 26 augusti 2010 07:27
>>> Till: FreeSWITCH Users Help; freeswitch-dev at lists.freeswitch.org
>>> ?mne: [Freeswitch-dev] Module written in C++
>>>
>>> I am trying to write a module having multiple C++ source files. When I
>>> run make, I find that only the C++ file having the name same as the module
>>> name is being compiled and all others do not produce any object file. As a
>>> result when I run FS I get linker error. Can anyone tell me how to get them
>>> compiled too?
>>>
>>> Thanks,
>>> Farhan
>>> !DSPAM:4c75fc8232934761421429!
>>>
>>> _______________________________________________
>>> 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
>>
>>
>> _______________________________________________
>> 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
>>
>
>
> _______________________________________________
> 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
>
>



-- 
Sincerely

Jay


From farhan.husain at csebuet.org  Thu Aug 26 22:43:13 2010
From: farhan.husain at csebuet.org (Farhan Husain)
Date: Fri, 27 Aug 2010 00:43:13 -0500
Subject: [Freeswitch-dev] Module written in C++
In-Reply-To: <AANLkTikOiJGO+gsBFAVWUJtO=xhPufXNX2nhTrva3Utq@mail.gmail.com>
References: <AANLkTinY_3rzkZfLeJOZ5WwPVyG4Ha9kciWYmXdfHTQA@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C57DC05816A@cooper>
	<AANLkTikgy7A7AP4saHiCZtwbzHYuzjCc+aMHB=8mquR6@mail.gmail.com>
	<AANLkTinE8WGxUDOS4mfzt5GxfEzYpLdc2zX=mCGxZ9Rg@mail.gmail.com>
	<AANLkTikOiJGO+gsBFAVWUJtO=xhPufXNX2nhTrva3Utq@mail.gmail.com>
Message-ID: <AANLkTinumxVLCfitm33x=xSuWfq3x3m51EuK1NKJU7v0@mail.gmail.com>

I added it in the wiki here:
http://wiki.freeswitch.org/wiki/Authoring_Freeswitch_Modules#Customizing_Include.2FLibrary_Flags

On Thu, Aug 26, 2010 at 6:19 PM, jay binks <jaybinks at gmail.com> wrote:

> no... dont keep it for YOUR archives..
>
> put it on the wiki..
> ( and post the url here please )
>
> J
>
>
> On Thu, Aug 26, 2010 at 4:23 PM, Steven Ayre <steveayre at gmail.com> wrote:
> > I'll just copy and paste that for the archives / my own records...
> >
> > BASE=../../../..
> > LOCAL_CFLAGS=-Wall -Werror
> > LOCAL_LDFLAGS=
> > LOCAL_OBJS=file.o file2.o file3.o
> >
> > local_depend: $(LOCAL_OBJS)
> >
> > include $(BASE)/build/modmake.rules
> >
> > -Steve
> >
> > On 26 August 2010 06:58, Farhan Husain <farhan.husain at csebuet.org>
> wrote:
> >>
> >> I just got the solution from Math at the IRC channel. Here is the link
> to
> >> the solution he gave me: http://pastebin.freeswitch.org/13727
> >>
> >> On Thu, Aug 26, 2010 at 12:36 AM, Peter Olsson
> >> <peter.olsson at visionutveckling.se> wrote:
> >>>
> >>> Have you updated the Makefile to build the other files?
> >>>
> >>> /Peter
> >>> ________________________________________
> >>> Fr?n: freeswitch-dev-bounces at lists.freeswitch.org
> >>> [freeswitch-dev-bounces at lists.freeswitch.org] f&#246;r Farhan Husain
> >>> [farhan.husain at csebuet.org]
> >>> Skickat: den 26 augusti 2010 07:27
> >>> Till: FreeSWITCH Users Help; freeswitch-dev at lists.freeswitch.org
> >>> ?mne: [Freeswitch-dev] Module written in C++
> >>>
> >>> I am trying to write a module having multiple C++ source files. When I
> >>> run make, I find that only the C++ file having the name same as the
> module
> >>> name is being compiled and all others do not produce any object file.
> As a
> >>> result when I run FS I get linker error. Can anyone tell me how to get
> them
> >>> compiled too?
> >>>
> >>> Thanks,
> >>> Farhan
> >>> !DSPAM:4c75fc8232934761421429!
> >>>
> >>> _______________________________________________
> >>> 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
> >>
> >>
> >> _______________________________________________
> >> 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
> >>
> >
> >
> > _______________________________________________
> > 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
> >
> >
>
>
>
> --
> Sincerely
>
> Jay
>
> _______________________________________________
> 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 farhan.husain at csebuet.org  Thu Aug 26 23:26:50 2010
From: farhan.husain at csebuet.org (Farhan Husain)
Date: Fri, 27 Aug 2010 01:26:50 -0500
Subject: [Freeswitch-dev] Module written in C++
In-Reply-To: <AANLkTinumxVLCfitm33x=xSuWfq3x3m51EuK1NKJU7v0@mail.gmail.com>
References: <AANLkTinY_3rzkZfLeJOZ5WwPVyG4Ha9kciWYmXdfHTQA@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C57DC05816A@cooper>
	<AANLkTikgy7A7AP4saHiCZtwbzHYuzjCc+aMHB=8mquR6@mail.gmail.com>
	<AANLkTinE8WGxUDOS4mfzt5GxfEzYpLdc2zX=mCGxZ9Rg@mail.gmail.com>
	<AANLkTikOiJGO+gsBFAVWUJtO=xhPufXNX2nhTrva3Utq@mail.gmail.com>
	<AANLkTinumxVLCfitm33x=xSuWfq3x3m51EuK1NKJU7v0@mail.gmail.com>
Message-ID: <AANLkTimQ1JTge5RyhVS2hpXw9XvJLtmHFP+jZKkruzKD@mail.gmail.com>

That place was not appropriate for the text so I moved it here:
http://wiki.freeswitch.org/wiki/Authoring_Freeswitch_Modules#Working_with_multiple_source_files

On Fri, Aug 27, 2010 at 12:43 AM, Farhan Husain
<farhan.husain at csebuet.org>wrote:

> I added it in the wiki here:
> http://wiki.freeswitch.org/wiki/Authoring_Freeswitch_Modules#Customizing_Include.2FLibrary_Flags
>
>
> On Thu, Aug 26, 2010 at 6:19 PM, jay binks <jaybinks at gmail.com> wrote:
>
>> no... dont keep it for YOUR archives..
>>
>> put it on the wiki..
>> ( and post the url here please )
>>
>> J
>>
>>
>> On Thu, Aug 26, 2010 at 4:23 PM, Steven Ayre <steveayre at gmail.com> wrote:
>> > I'll just copy and paste that for the archives / my own records...
>> >
>> > BASE=../../../..
>> > LOCAL_CFLAGS=-Wall -Werror
>> > LOCAL_LDFLAGS=
>> > LOCAL_OBJS=file.o file2.o file3.o
>> >
>> > local_depend: $(LOCAL_OBJS)
>> >
>> > include $(BASE)/build/modmake.rules
>> >
>> > -Steve
>> >
>> > On 26 August 2010 06:58, Farhan Husain <farhan.husain at csebuet.org>
>> wrote:
>> >>
>> >> I just got the solution from Math at the IRC channel. Here is the link
>> to
>> >> the solution he gave me: http://pastebin.freeswitch.org/13727
>> >>
>> >> On Thu, Aug 26, 2010 at 12:36 AM, Peter Olsson
>> >> <peter.olsson at visionutveckling.se> wrote:
>> >>>
>> >>> Have you updated the Makefile to build the other files?
>> >>>
>> >>> /Peter
>> >>> ________________________________________
>> >>> Fr?n: freeswitch-dev-bounces at lists.freeswitch.org
>> >>> [freeswitch-dev-bounces at lists.freeswitch.org] f&#246;r Farhan Husain
>> >>> [farhan.husain at csebuet.org]
>> >>> Skickat: den 26 augusti 2010 07:27
>> >>> Till: FreeSWITCH Users Help; freeswitch-dev at lists.freeswitch.org
>> >>> ?mne: [Freeswitch-dev] Module written in C++
>> >>>
>> >>> I am trying to write a module having multiple C++ source files. When I
>> >>> run make, I find that only the C++ file having the name same as the
>> module
>> >>> name is being compiled and all others do not produce any object file.
>> As a
>> >>> result when I run FS I get linker error. Can anyone tell me how to get
>> them
>> >>> compiled too?
>> >>>
>> >>> Thanks,
>> >>> Farhan
>> >>> !DSPAM:4c75fc8232934761421429!
>> >>>
>> >>> _______________________________________________
>> >>> 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
>> >>
>> >>
>> >> _______________________________________________
>> >> 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
>> >>
>> >
>> >
>> > _______________________________________________
>> > 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
>> >
>> >
>>
>>
>>
>> --
>> Sincerely
>>
>> Jay
>>
>> _______________________________________________
>> 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 jkr888 at gmail.com  Fri Aug 27 07:20:58 2010
From: jkr888 at gmail.com (Johny Kadarisman Kwan)
Date: Fri, 27 Aug 2010 10:20:58 -0400
Subject: [Freeswitch-dev] FS complained on RTP packet.
In-Reply-To: <48B36522-3EB7-408F-BD65-ED22BE972D07@jerris.com>
References: <AANLkTimztouRpHws8s8eOrWDkemUSBVrGG4kLpEEqqc+@mail.gmail.com>
	<4EBBAC6B-3476-4D36-81AA-96B6088B9614@jerris.com>
	<AANLkTikUan6sxDhwAyrUUpdRrrt9WAh=3dvgzYGxQ5Dn@mail.gmail.com>
	<AANLkTi=b3mB-SyhCEojmqMg7vVS1ULKL1-3cNQaVuZHB@mail.gmail.com>
	<EEA61F66-395B-4A53-8BF5-AEA111898FF4@freeswitch.org>
	<AANLkTi=PbPg+v5dsG_pJxrv24gEdWH_mK=STfpMkNuWf@mail.gmail.com>
	<48B36522-3EB7-408F-BD65-ED22BE972D07@jerris.com>
Message-ID: <AANLkTikn4CsoMTk4Oti_trKYAzaoSAT8Qvzt62yZqKQV@mail.gmail.com>

Is ptime adaptable? i won't know what is the rate at the beginning of
sip/sdp negotiation. But it possible to calculate how much audio once i
process the upstream audio. So, is it possible to change the ptime while
call in progress phase?

Btw, everything works fine now, with 20ms, 160 bytes chunk and FS just play
that smoothly. Thanks for all the pointer.

JK

On Thu, Aug 26, 2010 at 12:49 PM, Michael Jerris <mike at jerris.com> wrote:

> If you are already getting them in larger chunks, you might as well pass
> along the packet size you get instead of breaking them up, just make sure to
> set the ptime correct.  What is the source of the audio?
>
> Mike
>
> On Aug 26, 2010, at 10:15 AM, Johny Kadarisman Kwan wrote:
>
> I adjust rtp timestamp to increment += 160, still no good audio.
> i took code that handle speex previously, timestamp was set to
> 320increment. seems working fine with speex/16k
>
> still no good ulaw audio. I'm converting up stream audio that sent to me in
> a large chunk, do some processing and now breaking up into smaller rtp
> chunk. do i have to limit the rtp paket rate to freeswitch (ie. 20ms delay
> between them)?
>
> Thanks again.
>
> On Thu, Aug 26, 2010 at 10:02 AM, Brian West <brian at freeswitch.org> wrote:
>
>> Its could be your timestamps too... how many are you incrementing on each
>> time stamp?  If you lie about time timestamps say send timestamps that jump
>> by 320 but only send 160 byte payload you're still going to get the warning
>> I'm pretty sure.
>>
>> /b
>>
>> On Aug 26, 2010, at 8:54 AM, Johny Kadarisman Kwan wrote:
>>
>> > Its excellent that  FS able to auto adjust to 40ms.Sending 160 bytes
>> payload does eliminate warning message from FS.
>> > My audio doesn't work yet, problem must be something else. At least no
>> more issues on rtp audio framing ;)
>> >
>> > Thanks,
>> > JK
>>
>>
>> _______________________________________________
>> 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
>>
>
> _______________________________________________
> 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
>
>
>
> _______________________________________________
> 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 steveayre at gmail.com  Fri Aug 27 09:09:57 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 27 Aug 2010 17:09:57 +0100
Subject: [Freeswitch-dev] FS complained on RTP packet.
In-Reply-To: <AANLkTikn4CsoMTk4Oti_trKYAzaoSAT8Qvzt62yZqKQV@mail.gmail.com>
References: <AANLkTimztouRpHws8s8eOrWDkemUSBVrGG4kLpEEqqc+@mail.gmail.com>
	<4EBBAC6B-3476-4D36-81AA-96B6088B9614@jerris.com>
	<AANLkTikUan6sxDhwAyrUUpdRrrt9WAh=3dvgzYGxQ5Dn@mail.gmail.com>
	<AANLkTi=b3mB-SyhCEojmqMg7vVS1ULKL1-3cNQaVuZHB@mail.gmail.com>
	<EEA61F66-395B-4A53-8BF5-AEA111898FF4@freeswitch.org>
	<AANLkTi=PbPg+v5dsG_pJxrv24gEdWH_mK=STfpMkNuWf@mail.gmail.com>
	<48B36522-3EB7-408F-BD65-ED22BE972D07@jerris.com>
	<AANLkTikn4CsoMTk4Oti_trKYAzaoSAT8Qvzt62yZqKQV@mail.gmail.com>
Message-ID: <AANLkTi=9+are0CRZPsEKQTW_QHcH+53UNNR+ceFJr+ZR@mail.gmail.com>

AFAIK no, but UPDATE can renegotiate codecs so that might be able to adjust
ptime.

In practice, if you're a proxy you're not handling media so you pass through
the SDP unchanged. So ptime will be what the other end said it's sending and
you don't need to work about ptime being different, because you won't send
anything until you know what ptime you'll be using.

For a B2BUA like FreeSWITCH, it should be able to use different ptimes on
different legs since each leg is a separate media stream. Converting
20ms->40ms It should queue up 40ms and then send it, and for 40ms->20ms it
should send two 20ms packets for every 40ms packet. Sure you can't match leg
a's ptime to leg b's, but the ptime used is left up to the client and it
probably doesn't matter since it'll still work.

At least that's how I think it'd work out.

-Steve


On 27 August 2010 15:20, Johny Kadarisman Kwan <jkr888 at gmail.com> wrote:

> Is ptime adaptable? i won't know what is the rate at the beginning of
> sip/sdp negotiation. But it possible to calculate how much audio once i
> process the upstream audio. So, is it possible to change the ptime while
> call in progress phase?
>
> Btw, everything works fine now, with 20ms, 160 bytes chunk and FS just play
> that smoothly. Thanks for all the pointer.
>
> JK
>
>
> On Thu, Aug 26, 2010 at 12:49 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> If you are already getting them in larger chunks, you might as well pass
>> along the packet size you get instead of breaking them up, just make sure to
>> set the ptime correct.  What is the source of the audio?
>>
>> Mike
>>
>> On Aug 26, 2010, at 10:15 AM, Johny Kadarisman Kwan wrote:
>>
>> I adjust rtp timestamp to increment += 160, still no good audio.
>> i took code that handle speex previously, timestamp was set to
>> 320increment. seems working fine with speex/16k
>>
>> still no good ulaw audio. I'm converting up stream audio that sent to me
>> in a large chunk, do some processing and now breaking up into smaller rtp
>> chunk. do i have to limit the rtp paket rate to freeswitch (ie. 20ms delay
>> between them)?
>>
>> Thanks again.
>>
>> On Thu, Aug 26, 2010 at 10:02 AM, Brian West <brian at freeswitch.org>wrote:
>>
>>> Its could be your timestamps too... how many are you incrementing on each
>>> time stamp?  If you lie about time timestamps say send timestamps that jump
>>> by 320 but only send 160 byte payload you're still going to get the warning
>>> I'm pretty sure.
>>>
>>> /b
>>>
>>> On Aug 26, 2010, at 8:54 AM, Johny Kadarisman Kwan wrote:
>>>
>>> > Its excellent that  FS able to auto adjust to 40ms.Sending 160 bytes
>>> payload does eliminate warning message from FS.
>>> > My audio doesn't work yet, problem must be something else. At least no
>>> more issues on rtp audio framing ;)
>>> >
>>> > Thanks,
>>> > JK
>>>
>>>
>>> _______________________________________________
>>> 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
>>>
>>
>> _______________________________________________
>> 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
>>
>>
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> 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 mike at jerris.com  Fri Aug 27 09:28:46 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 27 Aug 2010 12:28:46 -0400
Subject: [Freeswitch-dev] FS complained on RTP packet.
In-Reply-To: <AANLkTi=9+are0CRZPsEKQTW_QHcH+53UNNR+ceFJr+ZR@mail.gmail.com>
References: <AANLkTimztouRpHws8s8eOrWDkemUSBVrGG4kLpEEqqc+@mail.gmail.com>
	<4EBBAC6B-3476-4D36-81AA-96B6088B9614@jerris.com>
	<AANLkTikUan6sxDhwAyrUUpdRrrt9WAh=3dvgzYGxQ5Dn@mail.gmail.com>
	<AANLkTi=b3mB-SyhCEojmqMg7vVS1ULKL1-3cNQaVuZHB@mail.gmail.com>
	<EEA61F66-395B-4A53-8BF5-AEA111898FF4@freeswitch.org>
	<AANLkTi=PbPg+v5dsG_pJxrv24gEdWH_mK=STfpMkNuWf@mail.gmail.com>
	<48B36522-3EB7-408F-BD65-ED22BE972D07@jerris.com>
	<AANLkTikn4CsoMTk4Oti_trKYAzaoSAT8Qvzt62yZqKQV@mail.gmail.com>
	<AANLkTi=9+are0CRZPsEKQTW_QHcH+53UNNR+ceFJr+ZR@mail.gmail.com>
Message-ID: <AE628F50-4653-41AC-8474-0E74AB6CE6BC@jerris.com>

The other end isn't sip and he ignored my question asking about what it actually is, so I was waiting on that response before providing an answer.  Its hard to provide a real answer when you only know 1/2 the problem.

On Aug 27, 2010, at 12:09 PM, Steven Ayre wrote:

> AFAIK no, but UPDATE can renegotiate codecs so that might be able to adjust ptime.
> 
> In practice, if you're a proxy you're not handling media so you pass through the SDP unchanged. So ptime will be what the other end said it's sending and you don't need to work about ptime being different, because you won't send anything until you know what ptime you'll be using.
> 
> For a B2BUA like FreeSWITCH, it should be able to use different ptimes on different legs since each leg is a separate media stream. Converting 20ms->40ms It should queue up 40ms and then send it, and for 40ms->20ms it should send two 20ms packets for every 40ms packet. Sure you can't match leg a's ptime to leg b's, but the ptime used is left up to the client and it probably doesn't matter since it'll still work.
> 
> At least that's how I think it'd work out.
> 
> -Steve
> 
> 
> On 27 August 2010 15:20, Johny Kadarisman Kwan <jkr888 at gmail.com> wrote:
> Is ptime adaptable? i won't know what is the rate at the beginning of sip/sdp negotiation. But it possible to calculate how much audio once i process the upstream audio. So, is it possible to change the ptime while call in progress phase?
> 
> Btw, everything works fine now, with 20ms, 160 bytes chunk and FS just play that smoothly. Thanks for all the pointer.
> 
> JK
> 
> 
> On Thu, Aug 26, 2010 at 12:49 PM, Michael Jerris <mike at jerris.com> wrote:
> If you are already getting them in larger chunks, you might as well pass along the packet size you get instead of breaking them up, just make sure to set the ptime correct.  What is the source of the audio?
> 
> Mike
> 
> On Aug 26, 2010, at 10:15 AM, Johny Kadarisman Kwan wrote:
> 
>> I adjust rtp timestamp to increment += 160, still no good audio. 
>> i took code that handle speex previously, timestamp was set to 320increment. seems working fine with speex/16k
>> 
>> still no good ulaw audio. I'm converting up stream audio that sent to me in a large chunk, do some processing and now breaking up into smaller rtp chunk. do i have to limit the rtp paket rate to freeswitch (ie. 20ms delay between them)?
>> 
>> Thanks again.
>> 
>> On Thu, Aug 26, 2010 at 10:02 AM, Brian West <brian at freeswitch.org> wrote:
>> Its could be your timestamps too... how many are you incrementing on each time stamp?  If you lie about time timestamps say send timestamps that jump by 320 but only send 160 byte payload you're still going to get the warning I'm pretty sure.
>> 
>> /b
>> 
>> On Aug 26, 2010, at 8:54 AM, Johny Kadarisman Kwan wrote:
>> 
>> > Its excellent that  FS able to auto adjust to 40ms.Sending 160 bytes payload does eliminate warning message from FS.
>> > My audio doesn't work yet, problem must be something else. At least no more issues on rtp audio framing ;)
>> >
>> > Thanks,
>> > JK
>> 

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From d at d-man.org  Fri Aug 27 14:44:29 2010
From: d at d-man.org (Darren Schreiber)
Date: Fri, 27 Aug 2010 14:44:29 -0700
Subject: [Freeswitch-dev] FreeSWITCH Training - Learn how to build a
	FreeSWITCH C module
Message-ID: <196C835FADC4D243A06AA032715AE72103FD@EXVMBX020-20.exch020.serverdata.net>

Hi folks,
After a very successful 1st training, I'm pleased to announce our second official FreeSWITCH Training in New York City!

The training is a 3-day bootcamp where we'll dive deep into FreeSWITCH and all the various goodies FreeSWITCH has to offer. Yes, even you, devs, might enjoy this training thoroughly and learn some new stuff.

Scheduled from October 13th-15th in downtown Manhattan, you'll learn everything you've ever wanted to know to master FreeSWITCH, including:

 *   Understanding configuration files and the default configuration
 *   Call authentication and routing basics
 *   Integration modules (mod_skypiax, mod_dingaling for Skype/GTalk/XMPP integration)
 *   the FreeSWITCH event system
 *   Load balancing and high availability
 *   FreeSWITCH Internals
 *   How to debug and troubleshoot FreeSWITCH
 *   Building Custom C Modules
 *   Advanced Modules
Registration is open now and group discounts are available. Please contact me for more information! Hope to see you in New York!

For more information, visit http://www.voipkb.com/

Sincerely,

Darren Schreiber
the 2600hz Project
www.2600hz.org<http://www.2600hz.org>


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From khovayko at gmail.com  Sun Aug 29 07:48:11 2010
From: khovayko at gmail.com (Oleg Khovayko)
Date: Sun, 29 Aug 2010 10:48:11 -0400
Subject: [Freeswitch-dev] FreeSWITCH reads commented lines from vars.xml
Message-ID: <4C7A732B.8070603@gmail.com>

Hi all,

Today I found bug in the FreeSWITCH v 1.0.6.

When I write in the file vars.xml following codecs config lines {{{

<X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA"/>

<!--
<X-PRE-PROCESS cmd="set" 
data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM,H264,H263,H263-1998"/> 

<X-PRE-PROCESS cmd="set" 
data="outbound_codec_prefs=PCMU,PCMA,GSM,H264,H263,H263-1998"/>
-->

}}}

FreeSWITCH reads config values from the comments, and, since comments 
located after correct values,
then values from comments overwrites correct config values.

and command: sofia status profile internal
shows:
CODECS IN               
G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM,H264,H263,H263-1998
CODECS OUT              
G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM,H264,H263,H263-1998

ATTN:
After modify vars.xml, for see this bug, need perform full restart of 
freeswitch, by commands like:
  /usr/local/etc/rc.d/freeswitch stop
/usr/local/etc/rc.d/freeswitch start

If you just execute "reloadxml" in the fs_cli, codecs lines will be 
unchanged.

Oleg



From gmaruzz at celliax.org  Sun Aug 29 08:05:06 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Sun, 29 Aug 2010 17:05:06 +0200
Subject: [Freeswitch-dev] FreeSWITCH reads commented lines from vars.xml
In-Reply-To: <4C7A732B.8070603@gmail.com>
References: <4C7A732B.8070603@gmail.com>
Message-ID: <AANLkTimE-FD3A-c2B9spW3HsGKN38OAL7J9dmS4mb2xh@mail.gmail.com>

On Sun, Aug 29, 2010 at 4:48 PM, Oleg Khovayko <khovayko at gmail.com> wrote:
> Hi all,
>
> Today I found bug in the FreeSWITCH v 1.0.6.
>
> When I write in the file vars.xml following codecs config lines {{{
>
> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA"/>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA"/>
>
> <!--
> <X-PRE-PROCESS cmd="set"
> data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM,H264,H263,H263-1998"/>
>
> <X-PRE-PROCESS cmd="set"
> data="outbound_codec_prefs=PCMU,PCMA,GSM,H264,H263,H263-1998"/>
> -->
>
> }}}
>
> FreeSWITCH reads config values from the comments, and, since comments
> located after correct values,
> then values from comments overwrites correct config values.


That's not a bug.

The PRE-PROCESS tags and values are read wherever they are.

They are not parsed by the XML parser that parse all the rest of the
config files, but they are used as macros by the preprocessor (that
runs before the XML parser).

So, if you want to comment it, you can edit as in:
X-COMMENTED-PRE-PROCESS so the preprocessor will not use it.

-giovanni


>
> and command: sofia status profile internal
> shows:
> CODECS IN
> G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM,H264,H263,H263-1998
> CODECS OUT
> G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM,H264,H263,H263-1998
>
> ATTN:
> After modify vars.xml, for see this bug, need perform full restart of
> freeswitch, by commands like:
> ?/usr/local/etc/rc.d/freeswitch stop
> /usr/local/etc/rc.d/freeswitch start
>
> If you just execute "reloadxml" in the fs_cli, codecs lines will be
> unchanged.
>
> Oleg
>
>
> _______________________________________________
> 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
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From dujinfang at gmail.com  Sun Aug 29 09:16:27 2010
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 30 Aug 2010 00:16:27 +0800
Subject: [Freeswitch-dev] FreeSWITCH reads commented lines from vars.xml
In-Reply-To: <AANLkTimE-FD3A-c2B9spW3HsGKN38OAL7J9dmS4mb2xh@mail.gmail.com>
References: <4C7A732B.8070603@gmail.com>
	<AANLkTimE-FD3A-c2B9spW3HsGKN38OAL7J9dmS4mb2xh@mail.gmail.com>
Message-ID: <AANLkTimDFa7F9+PXfb6N7uShtWFvYy7emMBmNByGw-GW@mail.gmail.com>

I use <xX-PRE-PROCESS   ;)

On Sun, Aug 29, 2010 at 11:05 PM, Giovanni Maruzzelli
<gmaruzz at celliax.org> wrote:
> On Sun, Aug 29, 2010 at 4:48 PM, Oleg Khovayko <khovayko at gmail.com> wrote:
>> Hi all,
>>
>> Today I found bug in the FreeSWITCH v 1.0.6.
>>
>> When I write in the file vars.xml following codecs config lines {{{
>>
>> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA"/>
>> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA"/>
>>
>> <!--
>> <X-PRE-PROCESS cmd="set"
>> data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM,H264,H263,H263-1998"/>
>>
>> <X-PRE-PROCESS cmd="set"
>> data="outbound_codec_prefs=PCMU,PCMA,GSM,H264,H263,H263-1998"/>
>> -->
>>
>> }}}
>>
>> FreeSWITCH reads config values from the comments, and, since comments
>> located after correct values,
>> then values from comments overwrites correct config values.
>
>
> That's not a bug.
>
> The PRE-PROCESS tags and values are read wherever they are.
>
> They are not parsed by the XML parser that parse all the rest of the
> config files, but they are used as macros by the preprocessor (that
> runs before the XML parser).
>
> So, if you want to comment it, you can edit as in:
> X-COMMENTED-PRE-PROCESS so the preprocessor will not use it.
>
> -giovanni
>
>
>>
>> and command: sofia status profile internal
>> shows:
>> CODECS IN
>> G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM,H264,H263,H263-1998
>> CODECS OUT
>> G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM,H264,H263,H263-1998
>>
>> ATTN:
>> After modify vars.xml, for see this bug, need perform full restart of
>> freeswitch, by commands like:
>> ?/usr/local/etc/rc.d/freeswitch stop
>> /usr/local/etc/rc.d/freeswitch start
>>
>> If you just execute "reloadxml" in the fs_cli, codecs lines will be
>> unchanged.
>>
>> Oleg
>>
>>
>> _______________________________________________
>> 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
>>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> 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
>



-- 
Blog: http://www.dujinfang.com
Proj:? http://www.freeswitch.org.cn


From msc at freeswitch.org  Sun Aug 29 19:44:23 2010
From: msc at freeswitch.org (Michael Collins)
Date: Sun, 29 Aug 2010 19:44:23 -0700
Subject: [Freeswitch-dev] FreeSWITCH reads commented lines from vars.xml
In-Reply-To: <AANLkTimE-FD3A-c2B9spW3HsGKN38OAL7J9dmS4mb2xh@mail.gmail.com>
References: <4C7A732B.8070603@gmail.com>
	<AANLkTimE-FD3A-c2B9spW3HsGKN38OAL7J9dmS4mb2xh@mail.gmail.com>
Message-ID: <AANLkTinGyf58REmhgS78zC=-SL9PSpoXofEiN7EOpujT@mail.gmail.com>

Giovanni,

This answer is very well-crafted. I think I will try to find a good spot on
the wiki to add this knowledge. I didn't see it explicitly mentioned, just
implicitly on the mod_xml_curl page. I'll add it to the list of stuff to
talk about on the next FS community conf call.

Thanks,
MC

On Sun, Aug 29, 2010 at 8:05 AM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:

> On Sun, Aug 29, 2010 at 4:48 PM, Oleg Khovayko <khovayko at gmail.com> wrote:
> > Hi all,
> >
> > Today I found bug in the FreeSWITCH v 1.0.6.
> >
> > When I write in the file vars.xml following codecs config lines {{{
> >
> > <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA"/>
> > <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA"/>
> >
> > <!--
> > <X-PRE-PROCESS cmd="set"
> > data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h
> ,G722,PCMU,PCMA,GSM,H264,H263,H263-1998"/>
> >
> > <X-PRE-PROCESS cmd="set"
> > data="outbound_codec_prefs=PCMU,PCMA,GSM,H264,H263,H263-1998"/>
> > -->
> >
> > }}}
> >
> > FreeSWITCH reads config values from the comments, and, since comments
> > located after correct values,
> > then values from comments overwrites correct config values.
>
>
> That's not a bug.
>
> The PRE-PROCESS tags and values are read wherever they are.
>
> They are not parsed by the XML parser that parse all the rest of the
> config files, but they are used as macros by the preprocessor (that
> runs before the XML parser).
>
> So, if you want to comment it, you can edit as in:
> X-COMMENTED-PRE-PROCESS so the preprocessor will not use it.
>
> -giovanni
>
>
> >
> > and command: sofia status profile internal
> > shows:
> > CODECS IN
> > G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM,H264,H263,H263-1998
> > CODECS OUT
> > G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM,H264,H263,H263-1998
> >
> > ATTN:
> > After modify vars.xml, for see this bug, need perform full restart of
> > freeswitch, by commands like:
> >  /usr/local/etc/rc.d/freeswitch stop
> > /usr/local/etc/rc.d/freeswitch start
> >
> > If you just execute "reloadxml" in the fs_cli, codecs lines will be
> > unchanged.
> >
> > Oleg
> >
> >
> > _______________________________________________
> > 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
> >
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> 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 rentmycoder at gmail.com  Sun Aug 29 23:42:25 2010
From: rentmycoder at gmail.com (rentmycoder rentmycoder)
Date: Mon, 30 Aug 2010 08:42:25 +0200
Subject: [Freeswitch-dev] RTMP support
Message-ID: <AANLkTin7H7Xp_FhEqxxrAj5akEcERHhePRDgk44oVbM0@mail.gmail.com>

Hi,

According to this node RTMP support is on the way:
http://www.freeswitch.org/node/277

Does anybody know any details about it?
Will it support video with transcoding or only audio with speex?


From mrene_lists at avgs.ca  Sun Aug 29 23:58:19 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Mon, 30 Aug 2010 02:58:19 -0400
Subject: [Freeswitch-dev] RTMP support
In-Reply-To: <AANLkTin7H7Xp_FhEqxxrAj5akEcERHhePRDgk44oVbM0@mail.gmail.com>
References: <AANLkTin7H7Xp_FhEqxxrAj5akEcERHhePRDgk44oVbM0@mail.gmail.com>
Message-ID: <2270BEF3-58C1-4B95-A450-BB2BF7DA2CD2@avgs.ca>

Hi,

Initial release will be audio only, I might throw in video soon enough since the additional implementation work isn't much.

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




On 2010-08-30, at 2:42 AM, rentmycoder rentmycoder wrote:

> Hi,
> 
> According to this node RTMP support is on the way:
> http://www.freeswitch.org/node/277
> 
> Does anybody know any details about it?
> Will it support video with transcoding or only audio with speex?
> 
> _______________________________________________
> 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 fdelawarde at wirelessmundi.com  Mon Aug 30 00:58:00 2010
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Mon, 30 Aug 2010 09:58:00 +0200
Subject: [Freeswitch-dev] RTMP support
In-Reply-To: <2270BEF3-58C1-4B95-A450-BB2BF7DA2CD2@avgs.ca>
References: <AANLkTin7H7Xp_FhEqxxrAj5akEcERHhePRDgk44oVbM0@mail.gmail.com>
	<2270BEF3-58C1-4B95-A450-BB2BF7DA2CD2@avgs.ca>
Message-ID: <1283155080.28748.32.camel@luna.tc.commsmundi.com>

That will be a really nice feature, thanks in advance! Will it need
anything else than FS (Red5 server, ...)?

Fran?ois.

On Mon, 2010-08-30 at 02:58 -0400, Mathieu Rene wrote:
> Hi,
> 
> Initial release will be audio only, I might throw in video soon enough since the additional implementation work isn't much.
> 
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
> 
> 
> 
> 
> On 2010-08-30, at 2:42 AM, rentmycoder rentmycoder wrote:
> 
> > Hi,
> > 
> > According to this node RTMP support is on the way:
> > http://www.freeswitch.org/node/277
> > 
> > Does anybody know any details about it?
> > Will it support video with transcoding or only audio with speex?
> > 
> > _______________________________________________
> > 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
> 
> 
> _______________________________________________
> 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 brian at freeswitch.org  Mon Aug 30 06:26:19 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 30 Aug 2010 08:26:19 -0500
Subject: [Freeswitch-dev] RTMP support
In-Reply-To: <1283155080.28748.32.camel@luna.tc.commsmundi.com>
References: <AANLkTin7H7Xp_FhEqxxrAj5akEcERHhePRDgk44oVbM0@mail.gmail.com>
	<2270BEF3-58C1-4B95-A450-BB2BF7DA2CD2@avgs.ca>
	<1283155080.28748.32.camel@luna.tc.commsmundi.com>
Message-ID: <D87D5F85-2491-44A9-8866-8661C6D7D012@freeswitch.org>

RED5 will not be required.

/b

On Aug 30, 2010, at 2:58 AM, Fran?ois Delawarde wrote:

> That will be a really nice feature, thanks in advance! Will it need
> anything else than FS (Red5 server, ...)?
> 
> Fran?ois.



From mustafa.pk at gmail.com  Mon Aug 30 06:32:30 2010
From: mustafa.pk at gmail.com (Ghulam Mustafa)
Date: Mon, 30 Aug 2010 18:32:30 +0500
Subject: [Freeswitch-dev] RTMP support
In-Reply-To: <D87D5F85-2491-44A9-8866-8661C6D7D012@freeswitch.org>
References: <AANLkTin7H7Xp_FhEqxxrAj5akEcERHhePRDgk44oVbM0@mail.gmail.com>
	<2270BEF3-58C1-4B95-A450-BB2BF7DA2CD2@avgs.ca>
	<1283155080.28748.32.camel@luna.tc.commsmundi.com>
	<D87D5F85-2491-44A9-8866-8661C6D7D012@freeswitch.org>
Message-ID: <AANLkTik+3hg7eNAMrUzf=580G1n2W-Dj97mMc5Mc=7wX@mail.gmail.com>

HI,

sorry for being naive, is it a rtmp client module or it would act as a rtmp
server asl well?

Regards,
-m

On Mon, Aug 30, 2010 at 6:26 PM, Brian West <brian at freeswitch.org> wrote:

> RED5 will not be required.
>
> /b
>
> On Aug 30, 2010, at 2:58 AM, Fran?ois Delawarde wrote:
>
> > That will be a really nice feature, thanks in advance! Will it need
> > anything else than FS (Red5 server, ...)?
> >
> > Fran?ois.
>
>
> _______________________________________________
> 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
>



-- 
Ghulam Mustafa
cell: +92 333.611.7681
sip: cyrenity at ekiga.net
mail: mustafa.pk at gmail.com
web: cyrenity.wordpress.com
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From mrene_lists at avgs.ca  Mon Aug 30 06:35:07 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Mon, 30 Aug 2010 09:35:07 -0400
Subject: [Freeswitch-dev] RTMP support
In-Reply-To: <AANLkTik+3hg7eNAMrUzf=580G1n2W-Dj97mMc5Mc=7wX@mail.gmail.com>
References: <AANLkTin7H7Xp_FhEqxxrAj5akEcERHhePRDgk44oVbM0@mail.gmail.com>
	<2270BEF3-58C1-4B95-A450-BB2BF7DA2CD2@avgs.ca>
	<1283155080.28748.32.camel@luna.tc.commsmundi.com>
	<D87D5F85-2491-44A9-8866-8661C6D7D012@freeswitch.org>
	<AANLkTik+3hg7eNAMrUzf=580G1n2W-Dj97mMc5Mc=7wX@mail.gmail.com>
Message-ID: <1F390F02-2EED-485E-A3BE-C706B6EEB924@avgs.ca>

Its a server module in the first place, with a call control API that can be called from flash.

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




On 2010-08-30, at 9:32 AM, Ghulam Mustafa wrote:

> HI, 
> 
> sorry for being naive, is it a rtmp client module or it would act as a rtmp server asl well? 
> 
> Regards,
> -m
> 
> On Mon, Aug 30, 2010 at 6:26 PM, Brian West <brian at freeswitch.org> wrote:
> RED5 will not be required.
> 
> /b
> 
> On Aug 30, 2010, at 2:58 AM, Fran?ois Delawarde wrote:
> 
> > That will be a really nice feature, thanks in advance! Will it need
> > anything else than FS (Red5 server, ...)?
> >
> > Fran?ois.
> 
> 
> _______________________________________________
> 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
> 
> 
> 
> -- 
> Ghulam Mustafa
> cell: +92 333.611.7681
> sip: cyrenity at ekiga.net
> mail: mustafa.pk at gmail.com
> web: cyrenity.wordpress.com
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org

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From mustafa.pk at gmail.com  Mon Aug 30 07:00:50 2010
From: mustafa.pk at gmail.com (Ghulam Mustafa)
Date: Mon, 30 Aug 2010 19:00:50 +0500
Subject: [Freeswitch-dev] RTMP support
In-Reply-To: <1F390F02-2EED-485E-A3BE-C706B6EEB924@avgs.ca>
References: <AANLkTin7H7Xp_FhEqxxrAj5akEcERHhePRDgk44oVbM0@mail.gmail.com>
	<2270BEF3-58C1-4B95-A450-BB2BF7DA2CD2@avgs.ca>
	<1283155080.28748.32.camel@luna.tc.commsmundi.com>
	<D87D5F85-2491-44A9-8866-8661C6D7D012@freeswitch.org>
	<AANLkTik+3hg7eNAMrUzf=580G1n2W-Dj97mMc5Mc=7wX@mail.gmail.com>
	<1F390F02-2EED-485E-A3BE-C706B6EEB924@avgs.ca>
Message-ID: <AANLkTinyJC3sQ7=1W_Dzz-6VYnXRF6PdUs_BOjY5vfEe@mail.gmail.com>

cute, so i can safely assume i would be able to create a flash client and
make rtmp -> sip calls and vice versa :)

-m

On Mon, Aug 30, 2010 at 6:35 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> Its a server module in the first place, with a call control API that can be
> called from flash.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 2010-08-30, at 9:32 AM, Ghulam Mustafa wrote:
>
> HI,
>
> sorry for being naive, is it a rtmp client module or it would act as a rtmp
> server asl well?
>
> Regards,
> -m
>
> On Mon, Aug 30, 2010 at 6:26 PM, Brian West <brian at freeswitch.org> wrote:
>
>> RED5 will not be required.
>>
>> /b
>>
>> On Aug 30, 2010, at 2:58 AM, Fran?ois Delawarde wrote:
>>
>> > That will be a really nice feature, thanks in advance! Will it need
>> > anything else than FS (Red5 server, ...)?
>> >
>> > Fran?ois.
>>
>>
>> _______________________________________________
>> 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
>>
>
>
>
> --
> Ghulam Mustafa
> cell: +92 333.611.7681
> sip: cyrenity at ekiga.net
> mail: mustafa.pk at gmail.com
> web: cyrenity.wordpress.com
>  _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> 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
>
>


-- 
Ghulam Mustafa
cell: +92 333.611.7681
sip: cyrenity at ekiga.net
mail: mustafa.pk at gmail.com
web: cyrenity.wordpress.com
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From matthew at matthew.at  Mon Aug 30 07:42:19 2010
From: matthew at matthew.at (Matthew Kaufman)
Date: Mon, 30 Aug 2010 07:42:19 -0700
Subject: [Freeswitch-dev] RTMP support
In-Reply-To: <2270BEF3-58C1-4B95-A450-BB2BF7DA2CD2@avgs.ca>
References: <AANLkTin7H7Xp_FhEqxxrAj5akEcERHhePRDgk44oVbM0@mail.gmail.com>
	<2270BEF3-58C1-4B95-A450-BB2BF7DA2CD2@avgs.ca>
Message-ID: <4C7BC34B.6000200@matthew.at>

  On 8/29/2010 11:58 PM, Mathieu Rene wrote:
> Hi,
>
> Initial release will be audio only, I might throw in video soon enough since the additional implementation work isn't much.
>
Just FYI, I've written several RTMP stacks myself (including one that 
interfaced with FreeSWITCH) and helped make the current public 
specification from Adobe much more correct than the first draft. I have 
some limitations as to the extent I can help (I was at Adobe when that 
spec was released, and I am now at Skype, and I wouldn't want to expose 
the project to any IP risks), but please drop me a note if something 
comes up and there's any chance I might be able to help.

Matthew Kaufman


From msc at freeswitch.org  Mon Aug 30 17:50:33 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 30 Aug 2010 17:50:33 -0700
Subject: [Freeswitch-dev] WANTED: Cookbook recipes
Message-ID: <AANLkTikZTySkoC9oNqD3w_+4BmYX5A+4p8pq6YtSRGFE@mail.gmail.com>

Hello all!

We are investigating the possibility of writing a cookbook for FreeSWITCH.
We would like to invite the community at large to submit ideas for recipes.
I have started a wiki page here:

http://wiki.freeswitch.org/wiki/Cookbook

Please add your cookbook wishlist items here. If you have an actual recipe
that you'd like to contribute then mention that fact as well. We are not
looking for the actual recipes themselves but rather the ideas for the
recipes. I've added a number of sections on the wiki but feel free to add
another one if I have missed a category of recipes.

Any questions please let me know!

Thanks,
MC
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From farhan.husain at csebuet.org  Mon Aug 30 22:47:15 2010
From: farhan.husain at csebuet.org (Farhan Husain)
Date: Tue, 31 Aug 2010 00:47:15 -0500
Subject: [Freeswitch-dev] How to get switch_core_session for a call
Message-ID: <AANLkTikUUu-CoaQPG-QKNkT7bUibhZXNaKwRPXDh0+qF@mail.gmail.com>

Hello,

Is there a way to get the switch_core_session of a call? Is it possible by
subscribing to any channel or any other event?

Thanks,
Farhan
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From mrene_lists at avgs.ca  Mon Aug 30 22:56:20 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Tue, 31 Aug 2010 01:56:20 -0400
Subject: [Freeswitch-dev] How to get switch_core_session for a call
In-Reply-To: <AANLkTikUUu-CoaQPG-QKNkT7bUibhZXNaKwRPXDh0+qF@mail.gmail.com>
References: <AANLkTikUUu-CoaQPG-QKNkT7bUibhZXNaKwRPXDh0+qF@mail.gmail.com>
Message-ID: <0F29D99E-A73C-4137-A4F8-A413E5107F78@avgs.ca>

Hi,

	switch_core_session_t *session;
	if ((session = switch_core_session_locate(uuid_here))) {
		/* do stuff with session */
		switch_core_session_rwunlock(session);
	}

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




On 2010-08-31, at 1:47 AM, Farhan Husain wrote:

> Hello,
> 
> Is there a way to get the switch_core_session of a call? Is it possible by subscribing to any channel or any other event?
> 
> Thanks,
> Farhan
> _______________________________________________
> 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 dujinfang at gmail.com  Tue Aug 31 02:34:46 2010
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 31 Aug 2010 17:34:46 +0800
Subject: [Freeswitch-dev] how to use switch_json?
Message-ID: <AANLkTineOswTuzr5UoLKA99Bbo6X7G24PyaEsuOFJ_q0@mail.gmail.com>

Hi,

I'm developing a client based on FSComm (using QT), I need to parse
some JSON string, as FS already has json libs build in so I don't want
link to other json libs.

I used cJSON_Parse, but when I tried to link with libfreeswitch, but
it cannot found cJSON_Parse

Undefined symbols:
  "_cJSON_Parse", referenced from:

I listed functions using

nm -gfj libfreeswitch.dylib (on mac), there are no json related

And there's no libjson in /usr/local/freeswitch/lib

So, where is it? As event socket support json format it should be somewhere.

I build mod_curl, then it download json lib again.

I'm on git version a few days ago.

Thanks.

-- 
Blog: http://www.dujinfang.com
Proj:? http://www.freeswitch.org.cn


From lists at haggett.org  Tue Aug 31 02:44:12 2010
From: lists at haggett.org (Thomas Haggett)
Date: Tue, 31 Aug 2010 10:44:12 +0100
Subject: [Freeswitch-dev] Issues building freeswitch (git-co) on Nexenta CP3
Message-ID: <E57D30D9-5BF4-4399-99FC-2A10685A9EAC@haggett.org>

Hi,

Having run away from OpenSolaris (but not too far) I've recently migrated to Nexenta and had a go at building / running FreeSWITCH. I run a few low-throughput SIP-based projects and have thus far been running Asterisk (is that a dirty word in these parts?) on a Debian VM which horribly exploded when I tried to build on nexenta. Not being much of a fan of Asterisk, I decided to give FS a go.

For reference, my git repo HEAD is at 6cdd3e2a2e4677b061b9fbb9ee516fae9601cd16.

It wasn't a clean build, per se, but I managed it with a couple of compile non-default switches and a slight hack that I figured I'd offer back for comment / improvement;

* configure guessed my platform as "i386-pc-solaris2.11" which caused a multitude of issues, principally bad linker behaviour meaning the libtool libraries failed with gcc complaining about bad file format (or something along those lines) for the generated .lo files half way through the build, as well as some atomic lock symbols missing from the c-library which I believe are emulated in libraries for i386 processors, but present on the amd64 host. (not an expert on these things). 

These were fixed (after a while picking through non-obvious errors) by simply adding the configure arg --build=amd64-pc-solaris5.11
Apologies if this is obvious to some, but it wasn't to me and took a while to figure out.

* a couple of modules wouldn't build cleanly but because I didn't require them I just commented them out from modules.conf. Trying to be useful I tried to build as many as I could and ended up with these modules removed:
#applications/mod_osp
#applications/mod_hash
#applications/mod_spandsp
#applications/mod_memcache
#codecs/mod_sangoma_codec
#codecs/mod_dahdi_codec
##directories/mod_ldap
#endpoints/mod_dingaling
##endpoints/mod_portaudio
#endpoints/mod_loopback
##endpoints/mod_alsa
##endpoints/mod_opal
##endpoints/mod_skinny
##endpoints/mod_skypopen
#endpoints/mod_h323
##../../libs/openzap/mod_openzap
##../../libs/freetdm/mod_freetdm
#asr_tts/mod_unimrcp
#asr_tts/mod_flite
#asr_tts/mod_pocketsphinx
#asr_tts/mod_cepstral
##asr_tts/mod_tts_commandline
#event_handlers/mod_erlang_event
#formats/mod_portaudio_stream
#languages/mod_python
#languages/mod_spidermonkey
#languages/mod_perl
#languages/mod_java
#languages/mod_managed
#xml_int/mod_xml_ldap
I believe a lot of these were simply missing 3rd party libraries but, since I didn't need them, I didn't spend much time working it out. I can do this if anyone is particularly interested.

* I had a couple of missing symbols (herror was one) which was simply resolved by adding LDFLAGS='-lresolv' to the configure command

* I had some issue with compiler warnings complaining about C-standards (don't have the error to hand), but I sorted this by simply adding CFLAGS='-fgnu89-inline', again to the configure.

* The only other fix was a missing definition for strcasecmp, which I struggled to sort the problem as <strings.h> was included in the appropriate files. I'm a little ashamed to say that I sorted this out by simply adding these library prototypes (lifted from strings.h) to libs/esl/src/include/esl.h:

extern int ffs(int);
extern int strcasecmp(const char *, const char *);
extern int strncasecmp(const char *, const char *, size_t);


After this the build proceeded (after quite some time) without issue and installed (and ran) fine. If anyone wants the specific errors in each case, I'll run through a build and try to reproduce and also file a bug if this is appropriate.

Now to figure out the Freeswitch configuration :)

Thomas.










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From rupa at rupa.com  Tue Aug 31 08:28:45 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 31 Aug 2010 10:28:45 -0500
Subject: [Freeswitch-dev] Issues building freeswitch (git-co) on Nexenta
	CP3
In-Reply-To: <E57D30D9-5BF4-4399-99FC-2A10685A9EAC@haggett.org>
References: <E57D30D9-5BF4-4399-99FC-2A10685A9EAC@haggett.org>
Message-ID: <AANLkTimQb=9wJ7=OcvZYhpQzAXvsDV6Qd_RkNophbhRf@mail.gmail.com>

On Tue, Aug 31, 2010 at 4:44 AM, Thomas Haggett <lists at haggett.org> wrote:

> * a couple of modules wouldn't build cleanly but because I didn't require
> them I just commented them out from modules.conf. Trying to be useful I
> tried to build as many as I could and ended up with these modules removed:
>
> #applications/mod_hash
> #applications/mod_spandsp
>
>
Both of these modules are pretty important.

spandsp gives you lots of codecs.

hash is used in the default dialplan.

I can't help with spandsp, but could you give me the errors you get when
mod_hash is included?

-- 
-Rupa
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From lists at haggett.org  Tue Aug 31 08:54:28 2010
From: lists at haggett.org (Thomas Haggett)
Date: Tue, 31 Aug 2010 16:54:28 +0100
Subject: [Freeswitch-dev] Issues building freeswitch (git-co) on Nexenta
	CP3
In-Reply-To: <AANLkTimQb=9wJ7=OcvZYhpQzAXvsDV6Qd_RkNophbhRf@mail.gmail.com>
References: <E57D30D9-5BF4-4399-99FC-2A10685A9EAC@haggett.org>
	<AANLkTimQb=9wJ7=OcvZYhpQzAXvsDV6Qd_RkNophbhRf@mail.gmail.com>
Message-ID: <86CF76E1-8D86-4885-AEC9-A2819F8697EC@haggett.org>

Hi,
Just re-compiled with hash enabled and got this build error (it was the C99 failure, it turns out);

making all mod_hash
In file included from /usr/include/time.h:35,
                 from /root/freeswitch/libs/esl/src/include/esl.h:159,
                 from src/esl.c:34:
/usr/include/sys/feature_tests.h:357:2: error: #error "Compiler or options invalid; UNIX 03 and POSIX.1-2001 applications       require the use of c99"
make[5]: *** [src/esl.o] Error 1
make[4]: *** [/root/freeswitch/libs/esl/libesl.so] Error 2
make[3]: *** [mod_hash-all] Error 1
make[2]: *** [all-recursive] Error 1

and mod_spandsp causes these errors;

cd /root/freeswitch/libs/spandsp && make -j1
make[1]: Entering directory `/root/freeswitch/libs/spandsp'
Making all in src
make[2]: Entering directory `/root/freeswitch/libs/spandsp/src'
make  all-am
make[3]: Entering directory `/root/freeswitch/libs/spandsp/src'
gcc -o make_modem_filter ../src/make_modem_filter.c ../src/filter_tools.c -DHAVE_CONFIG_H -I../src -lm
/var/tmp//ccpvaWbv.o: In function `make_tx_filter':
make_modem_filter.c:(.text+0x131): undefined reference to `__tgmath_fabs'
make_modem_filter.c:(.text+0x159): undefined reference to `__tgmath_fabs'
/var/tmp//ccpvaWbv.o: In function `make_rx_filter':
make_modem_filter.c:(.text+0x582): undefined reference to `__tgmath_fabs'
make_modem_filter.c:(.text+0x5aa): undefined reference to `__tgmath_fabs'
make_modem_filter.c:(.text+0x726): undefined reference to `__tgmath_cos'
make_modem_filter.c:(.text+0x761): undefined reference to `__tgmath_sin'
/var/tmp//ccqvaWbv.o: In function `expj':
filter_tools.c:(.text+0x3a5): undefined reference to `__tgmath_sin'
filter_tools.c:(.text+0x3bb): undefined reference to `__tgmath_cos'
/var/tmp//ccqvaWbv.o: In function `compute_raised_cosine_filter':
filter_tools.c:(.text+0x542): undefined reference to `__tgmath_cos'
filter_tools.c:(.text+0x634): undefined reference to `__tgmath_sqrt'
filter_tools.c:(.text+0x6a5): undefined reference to `__tgmath_sin'
/var/tmp//ccqvaWbv.o: In function `apply_hamming_window':
filter_tools.c:(.text+0x8b8): undefined reference to `__tgmath_cos'
/var/tmp//ccqvaWbv.o: In function `truncate_coeffs':
filter_tools.c:(.text+0x94b): undefined reference to `__tgmath_pow'
/var/tmp//ccqvaWbv.o: In function `fix':
filter_tools.c:(.text+0xa29): undefined reference to `__tgmath_floor'
filter_tools.c:(.text+0xa47): undefined reference to `__tgmath_floor'
collect2: ld returned 1 exit status
make[3]: *** [make_modem_filter] Error 1
make[3]: Leaving directory `/root/freeswitch/libs/spandsp/src'
make[2]: *** [all] Error 2
make[2]: Leaving directory `/root/freeswitch/libs/spandsp/src'
make[1]: *** [all-recursive] Error 1
make[1]: Leaving directory `/root/freeswitch/libs/spandsp'
make: *** [/root/freeswitch/libs/spandsp/src/libspandsp.la] Error 2

As I say, I have a working build without these modules and that seemed easier than investigating what header files / libraries were and were not present.

Cheers,
Thomas.

On 31 Aug 2010, at 16:28, Rupa Schomaker wrote:

> 
> 
> On Tue, Aug 31, 2010 at 4:44 AM, Thomas Haggett <lists at haggett.org> wrote:
> * a couple of modules wouldn't build cleanly but because I didn't require them I just commented them out from modules.conf. Trying to be useful I tried to build as many as I could and ended up with these modules removed:
> #applications/mod_hash
> #applications/mod_spandsp
> 
> 
> Both of these modules are pretty important.
> 
> spandsp gives you lots of codecs.
> 
> hash is used in the default dialplan.  
> 
> I can't help with spandsp, but could you give me the errors you get when mod_hash is included?
> 
> -- 
> -Rupa
> _______________________________________________
> 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  Tue Aug 31 09:04:30 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 31 Aug 2010 11:04:30 -0500
Subject: [Freeswitch-dev] Issues building freeswitch (git-co) on Nexenta
	CP3
In-Reply-To: <E57D30D9-5BF4-4399-99FC-2A10685A9EAC@haggett.org>
References: <E57D30D9-5BF4-4399-99FC-2A10685A9EAC@haggett.org>
Message-ID: <AANLkTinUFo8udYevEoQizBSxO8jJ=60tbkMJErJhNRzC@mail.gmail.com>

Asterisk is not a dirty word here.  Many people use it still.  Not my
favorite VoIP app but definitely a pioneer in the industry.


On Tue, Aug 31, 2010 at 4:44 AM, Thomas Haggett <lists at haggett.org> wrote:
> Hi,
> Having run away from OpenSolaris (but not too far) I've recently migrated to
> Nexenta and had a go at building / running FreeSWITCH. I run a few
> low-throughput SIP-based projects and have thus far been running Asterisk
> (is that a dirty word in these parts?) on a Debian VM which horribly
> exploded when I tried to build on nexenta. Not being much of a fan of
> Asterisk, I decided to give FS a go.
> For reference, my git repo HEAD is
> at?6cdd3e2a2e4677b061b9fbb9ee516fae9601cd16.
> It wasn't a clean build, per se, but I managed it with a couple of compile
> non-default switches and a slight hack that I figured I'd offer back for
> comment / improvement;
> * configure guessed my platform as "i386-pc-solaris2.11" which caused a
> multitude of issues, principally bad linker behaviour meaning the libtool
> libraries failed with gcc complaining about bad file format (or something
> along those lines) for the generated .lo files half way through the build,
> as well as some atomic lock symbols?missing?from the c-library which I
> believe are emulated in libraries for i386 processors, but present on the
> amd64 host. (not an expert on these things).
> These were fixed (after a while picking through non-obvious errors) by
> simply adding the configure arg --build=amd64-pc-solaris5.11
> Apologies if this is obvious to some, but it wasn't to me and took a while
> to figure out.
> * a couple of modules wouldn't build cleanly but because I didn't require
> them I just commented them out from modules.conf. Trying to be useful I
> tried to build as many as I could and ended up with these modules removed:
>
> #applications/mod_osp
> #applications/mod_hash
> #applications/mod_spandsp
> #applications/mod_memcache
> #codecs/mod_sangoma_codec
> #codecs/mod_dahdi_codec
> ##directories/mod_ldap
> #endpoints/mod_dingaling
> ##endpoints/mod_portaudio
> #endpoints/mod_loopback
> ##endpoints/mod_alsa
> ##endpoints/mod_opal
> ##endpoints/mod_skinny
> ##endpoints/mod_skypopen
> #endpoints/mod_h323
> ##../../libs/openzap/mod_openzap
> ##../../libs/freetdm/mod_freetdm
> #asr_tts/mod_unimrcp
> #asr_tts/mod_flite
> #asr_tts/mod_pocketsphinx
> #asr_tts/mod_cepstral
> ##asr_tts/mod_tts_commandline
> #event_handlers/mod_erlang_event
> #formats/mod_portaudio_stream
> #languages/mod_python
> #languages/mod_spidermonkey
> #languages/mod_perl
> #languages/mod_java
> #languages/mod_managed
> #xml_int/mod_xml_ldap
>
> I believe a lot of these were simply missing 3rd party libraries but, since
> I didn't need them, I didn't spend much time working it out. I can do this
> if anyone is particularly interested.
> * I had a couple of missing symbols (herror was one) which was simply
> resolved by adding LDFLAGS='-lresolv' to the configure command
> * I had some issue with compiler warnings complaining about C-standards
> (don't have the error to hand), but I sorted this by simply adding
> CFLAGS='-fgnu89-inline', again to the configure.
> * The only other fix was a missing definition for strcasecmp, which I
> struggled to sort the problem as <strings.h> was included in the appropriate
> files. I'm a little ashamed to say that I sorted this out by simply adding
> these library prototypes (lifted from strings.h) to
> libs/esl/src/include/esl.h:
>
> extern int ffs(int);
> extern int strcasecmp(const char *, const char *);
> extern int strncasecmp(const char *, const char *, size_t);
>
> After this the build proceeded (after quite some time) without issue and
> installed (and ran) fine. If anyone wants the specific errors in each case,
> I'll run through a build and try to reproduce and also file a bug if this is
> appropriate.
> Now to figure out the Freeswitch configuration :)
> Thomas.
>
>
>
>
>
>
>
>
>
>
> _______________________________________________
> 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 lists at haggett.org  Tue Aug 31 09:14:19 2010
From: lists at haggett.org (Thomas Haggett)
Date: Tue, 31 Aug 2010 17:14:19 +0100
Subject: [Freeswitch-dev] Issues building freeswitch (git-co) on Nexenta
	CP3
In-Reply-To: <AANLkTinUFo8udYevEoQizBSxO8jJ=60tbkMJErJhNRzC@mail.gmail.com>
References: <E57D30D9-5BF4-4399-99FC-2A10685A9EAC@haggett.org>
	<AANLkTinUFo8udYevEoQizBSxO8jJ=60tbkMJErJhNRzC@mail.gmail.com>
Message-ID: <67BBAAD8-4EE5-4442-94F6-0474A2489B61@haggett.org>

'twas a joke :) (not asterisk, the comment.)

I've had years of faithful service (albeit with small-ish projects) and lots of fun playing around with it's many features. Freeswitch does, however, feel much cleaner and seems to be much easier to manage, even with my limited experience of it thus far.

Apologies if anyone took offence (my British humour to blame, obviously...)

Thomas.

On 31 Aug 2010, at 17:04, Anthony Minessale wrote:

> Asterisk is not a dirty word here.  Many people use it still.  Not my
> favorite VoIP app but definitely a pioneer in the industry.
> 
> 
> On Tue, Aug 31, 2010 at 4:44 AM, Thomas Haggett <lists at haggett.org> wrote:
>> Hi,
>> Having run away from OpenSolaris (but not too far) I've recently migrated to
>> Nexenta and had a go at building / running FreeSWITCH. I run a few
>> low-throughput SIP-based projects and have thus far been running Asterisk
>> (is that a dirty word in these parts?) on a Debian VM which horribly
>> exploded when I tried to build on nexenta. Not being much of a fan of
>> Asterisk, I decided to give FS a go.
>> For reference, my git repo HEAD is
>> at 6cdd3e2a2e4677b061b9fbb9ee516fae9601cd16.
>> It wasn't a clean build, per se, but I managed it with a couple of compile
>> non-default switches and a slight hack that I figured I'd offer back for
>> comment / improvement;
>> * configure guessed my platform as "i386-pc-solaris2.11" which caused a
>> multitude of issues, principally bad linker behaviour meaning the libtool
>> libraries failed with gcc complaining about bad file format (or something
>> along those lines) for the generated .lo files half way through the build,
>> as well as some atomic lock symbols missing from the c-library which I
>> believe are emulated in libraries for i386 processors, but present on the
>> amd64 host. (not an expert on these things).
>> These were fixed (after a while picking through non-obvious errors) by
>> simply adding the configure arg --build=amd64-pc-solaris5.11
>> Apologies if this is obvious to some, but it wasn't to me and took a while
>> to figure out.
>> * a couple of modules wouldn't build cleanly but because I didn't require
>> them I just commented them out from modules.conf. Trying to be useful I
>> tried to build as many as I could and ended up with these modules removed:
>> 
>> #applications/mod_osp
>> #applications/mod_hash
>> #applications/mod_spandsp
>> #applications/mod_memcache
>> #codecs/mod_sangoma_codec
>> #codecs/mod_dahdi_codec
>> ##directories/mod_ldap
>> #endpoints/mod_dingaling
>> ##endpoints/mod_portaudio
>> #endpoints/mod_loopback
>> ##endpoints/mod_alsa
>> ##endpoints/mod_opal
>> ##endpoints/mod_skinny
>> ##endpoints/mod_skypopen
>> #endpoints/mod_h323
>> ##../../libs/openzap/mod_openzap
>> ##../../libs/freetdm/mod_freetdm
>> #asr_tts/mod_unimrcp
>> #asr_tts/mod_flite
>> #asr_tts/mod_pocketsphinx
>> #asr_tts/mod_cepstral
>> ##asr_tts/mod_tts_commandline
>> #event_handlers/mod_erlang_event
>> #formats/mod_portaudio_stream
>> #languages/mod_python
>> #languages/mod_spidermonkey
>> #languages/mod_perl
>> #languages/mod_java
>> #languages/mod_managed
>> #xml_int/mod_xml_ldap
>> 
>> I believe a lot of these were simply missing 3rd party libraries but, since
>> I didn't need them, I didn't spend much time working it out. I can do this
>> if anyone is particularly interested.
>> * I had a couple of missing symbols (herror was one) which was simply
>> resolved by adding LDFLAGS='-lresolv' to the configure command
>> * I had some issue with compiler warnings complaining about C-standards
>> (don't have the error to hand), but I sorted this by simply adding
>> CFLAGS='-fgnu89-inline', again to the configure.
>> * The only other fix was a missing definition for strcasecmp, which I
>> struggled to sort the problem as <strings.h> was included in the appropriate
>> files. I'm a little ashamed to say that I sorted this out by simply adding
>> these library prototypes (lifted from strings.h) to
>> libs/esl/src/include/esl.h:
>> 
>> extern int ffs(int);
>> extern int strcasecmp(const char *, const char *);
>> extern int strncasecmp(const char *, const char *, size_t);
>> 
>> After this the build proceeded (after quite some time) without issue and
>> installed (and ran) fine. If anyone wants the specific errors in each case,
>> I'll run through a build and try to reproduce and also file a bug if this is
>> appropriate.
>> Now to figure out the Freeswitch configuration :)
>> Thomas.
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> _______________________________________________
>> 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
> 
> _______________________________________________
> 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  Tue Aug 31 10:03:23 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 31 Aug 2010 12:03:23 -0500
Subject: [Freeswitch-dev] Issues building freeswitch (git-co) on Nexenta
	CP3
In-Reply-To: <67BBAAD8-4EE5-4442-94F6-0474A2489B61@haggett.org>
References: <E57D30D9-5BF4-4399-99FC-2A10685A9EAC@haggett.org>
	<AANLkTinUFo8udYevEoQizBSxO8jJ=60tbkMJErJhNRzC@mail.gmail.com>
	<67BBAAD8-4EE5-4442-94F6-0474A2489B61@haggett.org>
Message-ID: <AANLkTikHV_Fahf7EC=4gda=NmJ7Ti3ZdHppi4Pe7SLtU@mail.gmail.com>

I highly doubt anyone took offense =p
I just waned to be a good neighbor lol

However, I doubt you'll get the same sentiment if you ask their
mailing list about us =D



On Tue, Aug 31, 2010 at 11:14 AM, Thomas Haggett <lists at haggett.org> wrote:
> 'twas a joke :) (not asterisk, the comment.)
>
> I've had years of faithful service (albeit with small-ish projects) and lots of fun playing around with it's many features. Freeswitch does, however, feel much cleaner and seems to be much easier to manage, even with my limited experience of it thus far.
>
> Apologies if anyone took offence (my British humour to blame, obviously...)
>
> Thomas.
>
> On 31 Aug 2010, at 17:04, Anthony Minessale wrote:
>
>> Asterisk is not a dirty word here. ?Many people use it still. ?Not my
>> favorite VoIP app but definitely a pioneer in the industry.
>>
>>
>> On Tue, Aug 31, 2010 at 4:44 AM, Thomas Haggett <lists at haggett.org> wrote:
>>> Hi,
>>> Having run away from OpenSolaris (but not too far) I've recently migrated to
>>> Nexenta and had a go at building / running FreeSWITCH. I run a few
>>> low-throughput SIP-based projects and have thus far been running Asterisk
>>> (is that a dirty word in these parts?) on a Debian VM which horribly
>>> exploded when I tried to build on nexenta. Not being much of a fan of
>>> Asterisk, I decided to give FS a go.
>>> For reference, my git repo HEAD is
>>> at 6cdd3e2a2e4677b061b9fbb9ee516fae9601cd16.
>>> It wasn't a clean build, per se, but I managed it with a couple of compile
>>> non-default switches and a slight hack that I figured I'd offer back for
>>> comment / improvement;
>>> * configure guessed my platform as "i386-pc-solaris2.11" which caused a
>>> multitude of issues, principally bad linker behaviour meaning the libtool
>>> libraries failed with gcc complaining about bad file format (or something
>>> along those lines) for the generated .lo files half way through the build,
>>> as well as some atomic lock symbols missing from the c-library which I
>>> believe are emulated in libraries for i386 processors, but present on the
>>> amd64 host. (not an expert on these things).
>>> These were fixed (after a while picking through non-obvious errors) by
>>> simply adding the configure arg --build=amd64-pc-solaris5.11
>>> Apologies if this is obvious to some, but it wasn't to me and took a while
>>> to figure out.
>>> * a couple of modules wouldn't build cleanly but because I didn't require
>>> them I just commented them out from modules.conf. Trying to be useful I
>>> tried to build as many as I could and ended up with these modules removed:
>>>
>>> #applications/mod_osp
>>> #applications/mod_hash
>>> #applications/mod_spandsp
>>> #applications/mod_memcache
>>> #codecs/mod_sangoma_codec
>>> #codecs/mod_dahdi_codec
>>> ##directories/mod_ldap
>>> #endpoints/mod_dingaling
>>> ##endpoints/mod_portaudio
>>> #endpoints/mod_loopback
>>> ##endpoints/mod_alsa
>>> ##endpoints/mod_opal
>>> ##endpoints/mod_skinny
>>> ##endpoints/mod_skypopen
>>> #endpoints/mod_h323
>>> ##../../libs/openzap/mod_openzap
>>> ##../../libs/freetdm/mod_freetdm
>>> #asr_tts/mod_unimrcp
>>> #asr_tts/mod_flite
>>> #asr_tts/mod_pocketsphinx
>>> #asr_tts/mod_cepstral
>>> ##asr_tts/mod_tts_commandline
>>> #event_handlers/mod_erlang_event
>>> #formats/mod_portaudio_stream
>>> #languages/mod_python
>>> #languages/mod_spidermonkey
>>> #languages/mod_perl
>>> #languages/mod_java
>>> #languages/mod_managed
>>> #xml_int/mod_xml_ldap
>>>
>>> I believe a lot of these were simply missing 3rd party libraries but, since
>>> I didn't need them, I didn't spend much time working it out. I can do this
>>> if anyone is particularly interested.
>>> * I had a couple of missing symbols (herror was one) which was simply
>>> resolved by adding LDFLAGS='-lresolv' to the configure command
>>> * I had some issue with compiler warnings complaining about C-standards
>>> (don't have the error to hand), but I sorted this by simply adding
>>> CFLAGS='-fgnu89-inline', again to the configure.
>>> * The only other fix was a missing definition for strcasecmp, which I
>>> struggled to sort the problem as <strings.h> was included in the appropriate
>>> files. I'm a little ashamed to say that I sorted this out by simply adding
>>> these library prototypes (lifted from strings.h) to
>>> libs/esl/src/include/esl.h:
>>>
>>> extern int ffs(int);
>>> extern int strcasecmp(const char *, const char *);
>>> extern int strncasecmp(const char *, const char *, size_t);
>>>
>>> After this the build proceeded (after quite some time) without issue and
>>> installed (and ran) fine. If anyone wants the specific errors in each case,
>>> I'll run through a build and try to reproduce and also file a bug if this is
>>> appropriate.
>>> Now to figure out the Freeswitch configuration :)
>>> Thomas.
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> 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
>>
>> _______________________________________________
>> 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
>
>
> _______________________________________________
> 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 farhan.husain at csebuet.org  Tue Aug 31 10:18:50 2010
From: farhan.husain at csebuet.org (Farhan Husain)
Date: Tue, 31 Aug 2010 12:18:50 -0500
Subject: [Freeswitch-dev] How to get switch_core_session for a call
In-Reply-To: <0F29D99E-A73C-4137-A4F8-A413E5107F78@avgs.ca>
References: <AANLkTikUUu-CoaQPG-QKNkT7bUibhZXNaKwRPXDh0+qF@mail.gmail.com>
	<0F29D99E-A73C-4137-A4F8-A413E5107F78@avgs.ca>
Message-ID: <AANLkTi=ZOMQs1yC2Hmyh1YwANjFTGRT7pfr1nNj_iNe2@mail.gmail.com>

Thanks Mathieu.

On Tue, Aug 31, 2010 at 12:56 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> Hi,
>
>        switch_core_session_t *session;
>        if ((session = switch_core_session_locate(uuid_here))) {
>                /* do stuff with session */
>                switch_core_session_rwunlock(session);
>        }
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 2010-08-31, at 1:47 AM, Farhan Husain wrote:
>
> > Hello,
> >
> > Is there a way to get the switch_core_session of a call? Is it possible
> by subscribing to any channel or any other event?
> >
> > Thanks,
> > Farhan
> > _______________________________________________
> > 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
>
>
> _______________________________________________
> 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 rupa at rupa.com  Tue Aug 31 10:26:36 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 31 Aug 2010 12:26:36 -0500
Subject: [Freeswitch-dev] Issues building freeswitch (git-co) on Nexenta
	CP3
In-Reply-To: <86CF76E1-8D86-4885-AEC9-A2819F8697EC@haggett.org>
References: <E57D30D9-5BF4-4399-99FC-2A10685A9EAC@haggett.org>
	<AANLkTimQb=9wJ7=OcvZYhpQzAXvsDV6Qd_RkNophbhRf@mail.gmail.com>
	<86CF76E1-8D86-4885-AEC9-A2819F8697EC@haggett.org>
Message-ID: <AANLkTiknCTFj0UHZigmiDe9rBrjNFi3LVNVbpLeD9p_f@mail.gmail.com>

Ok, esl issue.  I'll pass on trying to fix that.

On Tue, Aug 31, 2010 at 10:54 AM, Thomas Haggett <lists at haggett.org> wrote:

> Just re-compiled with hash enabled and got this build error (it was the C99
> failure, it turns out);
>
> making all mod_hash
> In file included from /usr/include/time.h:35,
>                  from /root/freeswitch/libs/esl/src/include/esl.h:159,
>                  from src/esl.c:34:
> /usr/include/sys/feature_tests.h:357:2: error: #error "Compiler or options
> invalid; UNIX 03 and POSIX.1-2001 applications       require the use of c99"
> make[5]: *** [src/esl.o] Error 1
> make[4]: *** [/root/freeswitch/libs/esl/libesl.so] Error 2
> make[3]: *** [mod_hash-all] Error 1
> make[2]: *** [all-recursive] Error 1
>
>


-- 
-Rupa
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From gmaruzz at celliax.org  Tue Aug 31 11:02:28 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Tue, 31 Aug 2010 20:02:28 +0200
Subject: [Freeswitch-dev] How to get switch_core_session for a call
In-Reply-To: <0F29D99E-A73C-4137-A4F8-A413E5107F78@avgs.ca>
References: <AANLkTikUUu-CoaQPG-QKNkT7bUibhZXNaKwRPXDh0+qF@mail.gmail.com>
	<0F29D99E-A73C-4137-A4F8-A413E5107F78@avgs.ca>
Message-ID: <AANLkTimJsPFwAT636S-FyakbnVcCvW8YVDiZ8r=oHZND@mail.gmail.com>

On Tue, Aug 31, 2010 at 7:56 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
> Hi,
>
> ? ? ? ?switch_core_session_t *session;
> ? ? ? ?if ((session = switch_core_session_locate(uuid_here))) {
> ? ? ? ? ? ? ? ?/* do stuff with session */
> ? ? ? ? ? ? ? ?switch_core_session_rwunlock(session);

Just to stress the point: note that the
"switch_core_session_rwunlock(session);" is MANDATORY when you're done
with what you want do to with the session, because
"switch_core_session_locate" will lock it automatically.

-giovanni


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From gmaruzz at celliax.org  Tue Aug 31 11:02:28 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Tue, 31 Aug 2010 20:02:28 +0200
Subject: [Freeswitch-dev] How to get switch_core_session for a call
In-Reply-To: <0F29D99E-A73C-4137-A4F8-A413E5107F78@avgs.ca>
References: <AANLkTikUUu-CoaQPG-QKNkT7bUibhZXNaKwRPXDh0+qF@mail.gmail.com>
	<0F29D99E-A73C-4137-A4F8-A413E5107F78@avgs.ca>
Message-ID: <AANLkTimJsPFwAT636S-FyakbnVcCvW8YVDiZ8r=oHZND@mail.gmail.com>

On Tue, Aug 31, 2010 at 7:56 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
> Hi,
>
> ? ? ? ?switch_core_session_t *session;
> ? ? ? ?if ((session = switch_core_session_locate(uuid_here))) {
> ? ? ? ? ? ? ? ?/* do stuff with session */
> ? ? ? ? ? ? ? ?switch_core_session_rwunlock(session);

Just to stress the point: note that the
"switch_core_session_rwunlock(session);" is MANDATORY when you're done
with what you want do to with the session, because
"switch_core_session_locate" will lock it automatically.

-giovanni


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From gmaruzz at celliax.org  Tue Aug 31 11:02:28 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Tue, 31 Aug 2010 20:02:28 +0200
Subject: [Freeswitch-dev] How to get switch_core_session for a call
In-Reply-To: <0F29D99E-A73C-4137-A4F8-A413E5107F78@avgs.ca>
References: <AANLkTikUUu-CoaQPG-QKNkT7bUibhZXNaKwRPXDh0+qF@mail.gmail.com>
	<0F29D99E-A73C-4137-A4F8-A413E5107F78@avgs.ca>
Message-ID: <AANLkTimJsPFwAT636S-FyakbnVcCvW8YVDiZ8r=oHZND@mail.gmail.com>

On Tue, Aug 31, 2010 at 7:56 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
> Hi,
>
> ? ? ? ?switch_core_session_t *session;
> ? ? ? ?if ((session = switch_core_session_locate(uuid_here))) {
> ? ? ? ? ? ? ? ?/* do stuff with session */
> ? ? ? ? ? ? ? ?switch_core_session_rwunlock(session);

Just to stress the point: note that the
"switch_core_session_rwunlock(session);" is MANDATORY when you're done
with what you want do to with the session, because
"switch_core_session_locate" will lock it automatically.

-giovanni


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From jmesquita at freeswitch.org  Tue Aug 31 12:57:46 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Tue, 31 Aug 2010 16:57:46 -0300
Subject: [Freeswitch-dev] how to use switch_json?
In-Reply-To: <AANLkTineOswTuzr5UoLKA99Bbo6X7G24PyaEsuOFJ_q0@mail.gmail.com>
References: <AANLkTineOswTuzr5UoLKA99Bbo6X7G24PyaEsuOFJ_q0@mail.gmail.com>
Message-ID: <AANLkTi=Fer0DkWOvghFT4HrPFiS3wxEX4XmpieO0VXNB@mail.gmail.com>

Seven, out of curiosity, have you seen this:
http://qtwiki.org/Parsing_JSON_with_QT_using_standard_QT_library

Regards,
Jo?o Mesquita



On Tue, Aug 31, 2010 at 6:34 AM, Seven Du <dujinfang at gmail.com> wrote:

> Hi,
>
> I'm developing a client based on FSComm (using QT), I need to parse
> some JSON string, as FS already has json libs build in so I don't want
> link to other json libs.
>
> I used cJSON_Parse, but when I tried to link with libfreeswitch, but
> it cannot found cJSON_Parse
>
> Undefined symbols:
>  "_cJSON_Parse", referenced from:
>
> I listed functions using
>
> nm -gfj libfreeswitch.dylib (on mac), there are no json related
>
> And there's no libjson in /usr/local/freeswitch/lib
>
> So, where is it? As event socket support json format it should be
> somewhere.
>
> I build mod_curl, then it download json lib again.
>
> I'm on git version a few days ago.
>
> Thanks.
>
> --
> Blog: http://www.dujinfang.com
> Proj:  http://www.freeswitch.org.cn
>
> _______________________________________________
> 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  Tue Aug 31 13:16:05 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 31 Aug 2010 15:16:05 -0500
Subject: [Freeswitch-dev] how to use switch_json?
In-Reply-To: <AANLkTineOswTuzr5UoLKA99Bbo6X7G24PyaEsuOFJ_q0@mail.gmail.com>
References: <AANLkTineOswTuzr5UoLKA99Bbo6X7G24PyaEsuOFJ_q0@mail.gmail.com>
Message-ID: <AANLkTi=vWYMp-LWw_RwrMxXfTOBX1ZfbAAtR45u7fvpd@mail.gmail.com>

why are you basing it on FSCOMM instead of helping make the actual FSCOMM?

On Tue, Aug 31, 2010 at 4:34 AM, Seven Du <dujinfang at gmail.com> wrote:
> Hi,
>
> I'm developing a client based on FSComm (using QT), I need to parse
> some JSON string, as FS already has json libs build in so I don't want
> link to other json libs.
>
> I used cJSON_Parse, but when I tried to link with libfreeswitch, but
> it cannot found cJSON_Parse
>
> Undefined symbols:
> ?"_cJSON_Parse", referenced from:
>
> I listed functions using
>
> nm -gfj libfreeswitch.dylib (on mac), there are no json related
>
> And there's no libjson in /usr/local/freeswitch/lib
>
> So, where is it? As event socket support json format it should be somewhere.
>
> I build mod_curl, then it download json lib again.
>
> I'm on git version a few days ago.
>
> Thanks.
>
> --
> Blog: http://www.dujinfang.com
> Proj:? http://www.freeswitch.org.cn
>
> _______________________________________________
> 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 jmesquita at freeswitch.org  Tue Aug 31 13:28:10 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Tue, 31 Aug 2010 17:28:10 -0300
Subject: [Freeswitch-dev] how to use switch_json?
In-Reply-To: <AANLkTi=vWYMp-LWw_RwrMxXfTOBX1ZfbAAtR45u7fvpd@mail.gmail.com>
References: <AANLkTineOswTuzr5UoLKA99Bbo6X7G24PyaEsuOFJ_q0@mail.gmail.com>
	<AANLkTi=vWYMp-LWw_RwrMxXfTOBX1ZfbAAtR45u7fvpd@mail.gmail.com>
Message-ID: <AANLkTim1m3mH7p9=oczBwzU4z2Jet_SOih444V2qfCLt@mail.gmail.com>

I was clicking reply when I saw your email Tony. Took the words from my
mouth.

Regards,
Jo?o Mesquita



On Tue, Aug 31, 2010 at 5:16 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> why are you basing it on FSCOMM instead of helping make the actual FSCOMM?
>
> On Tue, Aug 31, 2010 at 4:34 AM, Seven Du <dujinfang at gmail.com> wrote:
> > Hi,
> >
> > I'm developing a client based on FSComm (using QT), I need to parse
> > some JSON string, as FS already has json libs build in so I don't want
> > link to other json libs.
> >
> > I used cJSON_Parse, but when I tried to link with libfreeswitch, but
> > it cannot found cJSON_Parse
> >
> > Undefined symbols:
> >  "_cJSON_Parse", referenced from:
> >
> > I listed functions using
> >
> > nm -gfj libfreeswitch.dylib (on mac), there are no json related
> >
> > And there's no libjson in /usr/local/freeswitch/lib
> >
> > So, where is it? As event socket support json format it should be
> somewhere.
> >
> > I build mod_curl, then it download json lib again.
> >
> > I'm on git version a few days ago.
> >
> > Thanks.
> >
> > --
> > Blog: http://www.dujinfang.com
> > Proj:  http://www.freeswitch.org.cn
> >
> > _______________________________________________
> > 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 <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org
>
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From dujinfang at gmail.com  Tue Aug 31 18:13:12 2010
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 1 Sep 2010 09:13:12 +0800
Subject: [Freeswitch-dev] how to use switch_json?
In-Reply-To: <AANLkTi=vWYMp-LWw_RwrMxXfTOBX1ZfbAAtR45u7fvpd@mail.gmail.com>
References: <AANLkTineOswTuzr5UoLKA99Bbo6X7G24PyaEsuOFJ_q0@mail.gmail.com>
	<AANLkTi=vWYMp-LWw_RwrMxXfTOBX1ZfbAAtR45u7fvpd@mail.gmail.com>
Message-ID: <AANLkTim2ydQ8zV3ab8PU8fAhgg7h5jXjCRUyiYfHsGgs@mail.gmail.com>

On Wed, Sep 1, 2010 at 4:16 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> why are you basing it on FSCOMM instead of helping make the actual FSCOMM?
>

Because I'm not only working on a SIP client but a client will
interaction with our system with flash, socket communication etc. SIP
is only part of it.

And, for FSComm, I do have sth. to discuss

1) Some of my patches already in tree.

2) I do made a branch a few days ago, fixed some bugs and changed UI.
While I have some idea to make it better, I'm new to QT, and you know
that it's a little hard to merge UI codes. So I'd like to keep on
working the branch until it looks better.

3) The hard part of FSComm is the UI, It's hard to agree an UI. So I
think the better way is to work out my branch and we can see what need
to merge back to the master tree.

4) I once had a branch (and contrib dir) on SVN, but didn't follow
after turn to git. So, I'd like to get an access again so I can push
my code. I cannot access fisheye right now, so cannot tell the repo
structure.

5) As the idea of FSComm is to use libfreeswitch, not the FS source,
so,, would it be better to split into a stand alone project? I guess
it would be easy to maintain.

i.e. I can build and link with libfreeswitch in
/usr/local/freeswitch/{include,lib} without the FS source.


@Jo?o, thank you to make FSComm happen.

I'm using qjson (http://qjson.sourceforge.net/) right now. It works
great, but as it need extra steps to link it and make it in release
package(especially for multi-OS), I'd like to find a "native"
solution.
I will try the QT script one. But at the same time, I still interested
to link to cJSON.

Thanks.

> On Tue, Aug 31, 2010 at 4:34 AM, Seven Du <dujinfang at gmail.com> wrote:
>> Hi,
>>
>> I'm developing a client based on FSComm (using QT), I need to parse
>> some JSON string, as FS already has json libs build in so I don't want
>> link to other json libs.
>>
>> I used cJSON_Parse, but when I tried to link with libfreeswitch, but
>> it cannot found cJSON_Parse
>>
>> Undefined symbols:
>> ?"_cJSON_Parse", referenced from:
>>
>> I listed functions using
>>
>> nm -gfj libfreeswitch.dylib (on mac), there are no json related
>>
>> And there's no libjson in /usr/local/freeswitch/lib
>>
>> So, where is it? As event socket support json format it should be somewhere.
>>
>> I build mod_curl, then it download json lib again.
>>
>> I'm on git version a few days ago.
>>
>> Thanks.
>>
>> --
>> Blog: http://www.dujinfang.com
>> Proj:? http://www.freeswitch.org.cn
>>
>> _______________________________________________
>> 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
>
> _______________________________________________
> 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
>



-- 
Blog: http://www.dujinfang.com
Proj:? http://www.freeswitch.org.cn


From dujinfang at gmail.com  Tue Aug 31 18:30:50 2010
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 1 Sep 2010 09:30:50 +0800
Subject: [Freeswitch-dev] how to use switch_json?
In-Reply-To: <AANLkTim2ydQ8zV3ab8PU8fAhgg7h5jXjCRUyiYfHsGgs@mail.gmail.com>
References: <AANLkTineOswTuzr5UoLKA99Bbo6X7G24PyaEsuOFJ_q0@mail.gmail.com>
	<AANLkTi=vWYMp-LWw_RwrMxXfTOBX1ZfbAAtR45u7fvpd@mail.gmail.com>
	<AANLkTim2ydQ8zV3ab8PU8fAhgg7h5jXjCRUyiYfHsGgs@mail.gmail.com>
Message-ID: <AANLkTima4qcJV+cBCHNX2KhW+G7oEH8hBJi+z08LL5vU@mail.gmail.com>

Got it, cJSON in libfreeswitch.a but not .dylib, how does FS use it?

On Wed, Sep 1, 2010 at 9:13 AM, Seven Du <dujinfang at gmail.com> wrote:
> On Wed, Sep 1, 2010 at 4:16 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> why are you basing it on FSCOMM instead of helping make the actual FSCOMM?
>>
>
> Because I'm not only working on a SIP client but a client will
> interaction with our system with flash, socket communication etc. SIP
> is only part of it.
>
> And, for FSComm, I do have sth. to discuss
>
> 1) Some of my patches already in tree.
>
> 2) I do made a branch a few days ago, fixed some bugs and changed UI.
> While I have some idea to make it better, I'm new to QT, and you know
> that it's a little hard to merge UI codes. So I'd like to keep on
> working the branch until it looks better.
>
> 3) The hard part of FSComm is the UI, It's hard to agree an UI. So I
> think the better way is to work out my branch and we can see what need
> to merge back to the master tree.
>
> 4) I once had a branch (and contrib dir) on SVN, but didn't follow
> after turn to git. So, I'd like to get an access again so I can push
> my code. I cannot access fisheye right now, so cannot tell the repo
> structure.
>
> 5) As the idea of FSComm is to use libfreeswitch, not the FS source,
> so,, would it be better to split into a stand alone project? I guess
> it would be easy to maintain.
>
> i.e. I can build and link with libfreeswitch in
> /usr/local/freeswitch/{include,lib} without the FS source.
>
>
> @Jo?o, thank you to make FSComm happen.
>
> I'm using qjson (http://qjson.sourceforge.net/) right now. It works
> great, but as it need extra steps to link it and make it in release
> package(especially for multi-OS), I'd like to find a "native"
> solution.
> I will try the QT script one. But at the same time, I still interested
> to link to cJSON.
>
> Thanks.
>
>> On Tue, Aug 31, 2010 at 4:34 AM, Seven Du <dujinfang at gmail.com> wrote:
>>> Hi,
>>>
>>> I'm developing a client based on FSComm (using QT), I need to parse
>>> some JSON string, as FS already has json libs build in so I don't want
>>> link to other json libs.
>>>
>>> I used cJSON_Parse, but when I tried to link with libfreeswitch, but
>>> it cannot found cJSON_Parse
>>>
>>> Undefined symbols:
>>> ?"_cJSON_Parse", referenced from:
>>>
>>> I listed functions using
>>>
>>> nm -gfj libfreeswitch.dylib (on mac), there are no json related
>>>
>>> And there's no libjson in /usr/local/freeswitch/lib
>>>
>>> So, where is it? As event socket support json format it should be somewhere.
>>>
>>> I build mod_curl, then it download json lib again.
>>>
>>> I'm on git version a few days ago.
>>>
>>> Thanks.
>>>
>>> --
>>> Blog: http://www.dujinfang.com
>>> Proj:? http://www.freeswitch.org.cn
>>>
>>> _______________________________________________
>>> 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
>>
>> _______________________________________________
>> 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
>>
>
>
>
> --
> Blog: http://www.dujinfang.com
> Proj:? http://www.freeswitch.org.cn
>



-- 
Blog: http://www.dujinfang.com
Proj:? http://www.freeswitch.org.cn