From vfclists at googlemail.com  Sat May  1 00:51:25 2010
From: vfclists at googlemail.com (Frank Church)
Date: Sat, 1 May 2010 08:51:25 +0100
Subject: [Freeswitch-users] mod_xml_cdr is not logging at all.
Message-ID: <n2z7abab2411005010051ye18ea1f2s5e5cf2e57158d09c@mail.gmail.com>

Mod_xml_cdr is not logging anything at all. I have created an additional
directory called logs, and it is still not working.
cdr_csv is logging fine.

Below is the current file, are there any faults in it?
Which directory does {prefix} point to?

My current verson is 17408m on Windows

<configuration name="xml_cdr.conf" description="XML CDR CURL logger">
>   <settings>
>     <param name="url" value="http://192.168.1.20:8101/"/>
>     <param name="log-dir" value="log/xml_cdr"/>
>     <param name="log-b-leg" value="false"/>
>     <param name="prefix-a-leg" value="true"/>
>     <param name="encode" value="true"/>
>      <param name="err-log-dir" value="log/xml_cdr"/> -->
>   </settings>
> </configuration>
>


-- 
Frank Church

=======================
http://devblog.brahmancreations.com
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From stevendt at primrosebank.net  Sat May  1 01:23:36 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Sat, 1 May 2010 09:23:36 +0100
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 46, Issue 347
References: <mailman.2546.1272641691.19150.freeswitch-users@lists.freeswitch.org>
	<z2g7b463d951004301920k5fc7f24aif66b0f997a71db6@mail.gmail.com>
Message-ID: <DCA0DEE74677499EB831E120CB2E6075@bp1.ad.bp.com>

Hi,

thanks a lot for the pointer. I had a copy (another version) of the Admin 
guide, but I have downloaded this one too - thanks.

As you say, the guide does list the options, but there's not quite enough 
(at least for me) description of what they actually mean.

I'm going to give Brian's tip a try, but I'm "googling" to see if I can find 
out what I'm doing when choosing of option over another,

regards
Dave



----- Original Message ----- 
From: "guru singh" <grsingh750 at gmail.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, May 01, 2010 3:20 AM
Subject: Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 46, Issue 347


>
> From: "Dave Stevenson" <stevendt at primrosebank.net>
> To: <freeswitch-users at lists.freeswitch.org>
> Date: Fri, 30 Apr 2010 15:53:26 +0100
> Subject: Re: [Freeswitch-users] Picking up voicemail
> Hi Brian,
>
> thanks for the pointer.
>
> The SPA-3012 is pretty strong on configuration options, but the 
> documentation is very light on what they actually mean. Just so that I 
> know, what is AVT actually doing ?
>
> regards
> Dave
DTMF Tx Method               Select the method to transmit DTMF
signals to the far end:
                                           InBand, AVT, INFO, Auto,
InBand+INFO, or AVT+INFO.
                                           InBand sends DTMF using the
audio path. AVT sends
                                           DTMF as eypents. INFO uses
the SIP INFO method. Auto
                                           uses InBand or AVT based on
the outcome of codec negotiation.
                                           The default is Auto.

Shamelessly pasted from
http://www.cisco.com/en/US/docs/voice_ip_comm/csbpvga/ata/administration/guide/ATA_AG_v3_NC-WEB.pdf.
I was reading this manual when I saw your post. You should check it
out if you haven't already.

gs

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



From aep.lists at it46.se  Sat May  1 05:08:50 2010
From: aep.lists at it46.se (Alberto Escudero)
Date: Sat, 1 May 2010 14:08:50 +0200
Subject: [Freeswitch-users] event after hangup (Javascript)
In-Reply-To: <b7818a7faca042bfd028f33900ed0d0d.squirrel@correo.nodo50.org>
References: <b7818a7faca042bfd028f33900ed0d0d.squirrel@correo.nodo50.org>
Message-ID: <2db510c7de4905454ad17e9c8c322d30.squirrel@correo.nodo50.org>

Answering myself here in case others have experienced the same problem

If session. variables are accessed after hangup inside of the on_hangup()
javascript will exist quietly without any warning. In order to be able to
trigger an event (read here ESL) inside of on_hangup it is needed to cache
the value of any session.* variables.

.aep
-- 
Stopping junk mailers is good for the environment

> Hi,
>
> Is there a way to trigger a customized event after the user hangup the
> call?
>
> Although i define the
> function on_hangup(hup_session, how)
>
> it seems i can not trigger and event inside of it although i can run
> logging.
>
> Hints are welcome.
>
> -aep
> --
> Stopping junk mailers is good for the environment
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>




From pjintheusa at gmail.com  Sat May  1 06:12:23 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Sat, 1 May 2010 09:12:23 -0400
Subject: [Freeswitch-users] run error after building in vs 2008!
In-Reply-To: <i2v1fdf92721004302253l7dbdf254p1ff2b3884ec5a4d4@mail.gmail.com>
References: <i2v1fdf92721004302253l7dbdf254p1ff2b3884ec5a4d4@mail.gmail.com>
Message-ID: <l2o367751821005010612j46281f6fg17629c38bcd07f4@mail.gmail.com>

Do you have WS2_32.dll present in C:\Windows\System32 ???

I am not sure how your PC would run without it though.


On Sat, May 1, 2010 at 1:53 AM, babak yakhchali
<babak.freeswitch at gmail.com>wrote:

> Hi
> freeswitch is built without any problems but when running it gives the
> error about "entry point inet_ntop could not be located in dynamic link
> library WS2_32.dll"
> I'm using xp sp3.
> Thanks
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From vetali100 at gmail.com  Sat May  1 06:51:03 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Sat, 1 May 2010 16:51:03 +0300
Subject: [Freeswitch-users] How to get GMT date/time in channel variable
	"start_stamp", instead of local
Message-ID: <p2ra858e7691005010651m4fde4356z70f679b1294f711d@mail.gmail.com>

Hi,

Right now channel variable "start_stamp" represents *local *date/time (to
remind, it contains value in following format: 2010-05-01 16:31:58)

Is there a similar variable (like start_stamp_gmt?) that will provide
*GMT*formatted date/time?

Or maybe there are some settings that will make FreeSWITCH provide all date
/ time formatted variables in GMT instead of local.

Thank you,
Vitalie
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From infos at madovsky.org  Sat May  1 11:23:30 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 1 May 2010 14:23:30 -0400
Subject: [Freeswitch-users] fs_cli
References: <7B847F47C4BE42019B25DE706841F10B@MOBILEE1705><4BDAE9AF.3020602@ewetel.de>
	<x2x191c3a031004300832u78ede026k9478dbf20926b237@mail.gmail.com>
Message-ID: <0596673499F343A2B16902BCEB40A03E@MOBILEE1705>

GRrrrr, right, event socket was commented out....

Thanks !

  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, April 30, 2010 11:32 AM
  Subject: Re: [Freeswitch-users] fs_cli


  or load mod_event_socket ?


  On Fri, Apr 30, 2010 at 9:31 AM, Helmut Kuper <helmut.kuper at ewetel.de> wrote:

    -----BEGIN PGP SIGNED MESSAGE-----
    Hash: SHA1

    Hi,

    by default fs_cli connects to 127.0.0.1 (localhost).

    I guess you hit either the access list configured in acl.conf.xml and
    event_socket.conf.xml or you just have to add the -H parameter (for
    host) to fs_cli.

    regards
    helmut
    -----BEGIN PGP SIGNATURE-----
    Version: GnuPG v1.4.7 (MingW32)

    iD8DBQFL2umv4tZeNddg3dwRAtf5AKCVVZ4N8pXZxsUCAvvj84plLfBybwCeKgWe
    5AZhTFAUGLbLwf5JR9j0hhQ=
    =mx8U
    -----END PGP SIGNATURE-----

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




  -- 
  Anthony Minessale II

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

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

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



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


  _______________________________________________
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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From babak.freeswitch at gmail.com  Sat May  1 11:31:32 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Sat, 1 May 2010 23:01:32 +0430
Subject: [Freeswitch-users] run error after building in vs 2008!
In-Reply-To: <l2o367751821005010612j46281f6fg17629c38bcd07f4@mail.gmail.com>
References: <i2v1fdf92721004302253l7dbdf254p1ff2b3884ec5a4d4@mail.gmail.com>
	<l2o367751821005010612j46281f6fg17629c38bcd07f4@mail.gmail.com>
Message-ID: <n2i1fdf92721005011131o221b6dd4sf483e00613963d5@mail.gmail.com>

ya I've got that in place
but as I searched the web they say it's related to windows version (xp vista
. . .)
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From infos at madovsky.org  Sat May  1 12:12:58 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 1 May 2010 15:12:58 -0400
Subject: [Freeswitch-users] get a special var from accepted call
Message-ID: <DB9E87BCD48242F9A6C5BE41A912D970@MOBILEE1705>

Hi,

I'd like to set a var (trunk ID)  in the bridge that I can get from my SIP phone when the call is accepted.
channel variables in set / export ?

Thanks

Franck
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From david.ponzone at gmail.com  Sat May  1 12:24:57 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Sat, 1 May 2010 21:24:57 +0200
Subject: [Freeswitch-users] get a special var from accepted call
In-Reply-To: <DB9E87BCD48242F9A6C5BE41A912D970@MOBILEE1705>
References: <DB9E87BCD48242F9A6C5BE41A912D970@MOBILEE1705>
Message-ID: <A57E04F6-D0D1-4129-9174-96C98752B655@gmail.com>

Franck,

sorry, can you rephrase that ?

You want to set a variable in the dialplan that the called phone will  
receive ?

export is your friend then.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 01/05/2010 ? 21:12, Madovsky a ?crit :

> Hi,
>
> I'd like to set a var (trunk ID)  in the bridge that I can get from  
> my SIP phone when the call is accepted.
> channel variables in set / export ?
>
> Thanks
>
> Franck
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From pjintheusa at gmail.com  Sat May  1 12:29:59 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Sat, 1 May 2010 15:29:59 -0400
Subject: [Freeswitch-users] run error after building in vs 2008!
In-Reply-To: <n2i1fdf92721005011131o221b6dd4sf483e00613963d5@mail.gmail.com>
References: <i2v1fdf92721004302253l7dbdf254p1ff2b3884ec5a4d4@mail.gmail.com>
	<l2o367751821005010612j46281f6fg17629c38bcd07f4@mail.gmail.com>
	<n2i1fdf92721005011131o221b6dd4sf483e00613963d5@mail.gmail.com>
Message-ID: <x2z367751821005011229g3b6c217kbda766daac32a44d@mail.gmail.com>

Oh yeah - looks like that call is not supported in older os like XP.

http://social.msdn.microsoft.com/Forums/en-US/vcgeneral/thread/e40465f2-41b7-4243-ad33-15ae9366f4e6

FreeSWITCH does support XP though -
http://wiki.freeswitch.org/wiki/Users_Guide_Introduction

So I don't know.

On Sat, May 1, 2010 at 2:31 PM, babak yakhchali
<babak.freeswitch at gmail.com>wrote:

> ya I've got that in place
> but as I searched the web they say it's related to windows version (xp
> vista . . .)
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Sat May  1 12:32:12 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 1 May 2010 15:32:12 -0400
Subject: [Freeswitch-users] get a special var from accepted call
References: <DB9E87BCD48242F9A6C5BE41A912D970@MOBILEE1705>
	<A57E04F6-D0D1-4129-9174-96C98752B655@gmail.com>
Message-ID: <4BAB40A76D8B4748A3A562FC8A37009F@MOBILEE1705>

Hi David,

no, set a var in the dialplan that the caller phone will receive (I use mod_distributor to load balance trunks)

Thanks
  ----- Original Message ----- 
  From: David Ponzone 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Saturday, May 01, 2010 3:24 PM
  Subject: Re: [Freeswitch-users] get a special var from accepted call


  Franck,


  sorry, can you rephrase that ?


  You want to set a variable in the dialplan that the called phone will receive ?


  export is your friend then.


  David Ponzone  Direction Technique
  email: david.ponzone at ipeva.fr
  tel:      01 74 03 18 97
  gsm:   06 66 98 76 34


  Service Client IPeva
  tel:      0811 46 26 26
  www.ipeva.fr  -   www.ipeva-studio.com


  Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.








  Le 01/05/2010 ? 21:12, Madovsky a ?crit :


    Hi,

    I'd like to set a var (trunk ID)  in the bridge that I can get from my SIP phone when the call is accepted.
    channel variables in set / export ?

    Thanks

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





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


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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From oseslija at gmail.com  Sat May  1 12:34:45 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Sat, 1 May 2010 21:34:45 +0200
Subject: [Freeswitch-users] get a special var from accepted call
In-Reply-To: <DB9E87BCD48242F9A6C5BE41A912D970@MOBILEE1705>
References: <DB9E87BCD48242F9A6C5BE41A912D970@MOBILEE1705>
Message-ID: <y2l4468a6771005011234s20ce9d39hc286b5b74b5eb281@mail.gmail.com>

<action application="set" data="sip_rh_Foo"/>

will set Foo on 200 OK received time.

O.

On Sat, May 1, 2010 at 9:12 PM, Madovsky <infos at madovsky.org> wrote:

>  Hi,
>
> I'd like to set a var (trunk ID)  in the bridge that I can get from my SIP
> phone when the call is accepted.
> channel variables in set / export ?
>
> Thanks
>
> Franck
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From oseslija at gmail.com  Sat May  1 12:39:45 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Sat, 1 May 2010 21:39:45 +0200
Subject: [Freeswitch-users] get a special var from accepted call
In-Reply-To: <4BAB40A76D8B4748A3A562FC8A37009F@MOBILEE1705>
References: <DB9E87BCD48242F9A6C5BE41A912D970@MOBILEE1705>
	<A57E04F6-D0D1-4129-9174-96C98752B655@gmail.com>
	<4BAB40A76D8B4748A3A562FC8A37009F@MOBILEE1705>
Message-ID: <s2w4468a6771005011239o26875bceza0397e708cab444b@mail.gmail.com>

You need to send a certain SIP header/value pair?


On Sat, May 1, 2010 at 9:32 PM, Madovsky <infos at madovsky.org> wrote:

>  Hi David,
>
> no, set a var in the dialplan that the caller phone will receive (I use
> mod_distributor to load balance trunks)
>
> Thanks
>
> ----- Original Message -----
> *From:* David Ponzone <david.ponzone at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Saturday, May 01, 2010 3:24 PM
> *Subject:* Re: [Freeswitch-users] get a special var from accepted call
>
> Franck,
>
> sorry, can you rephrase that ?
>
> You want to set a variable in the dialplan that the called phone will
> receive ?
>
> export is your friend then.
>
>   David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
>  tel:      0811 46 26 26
>  www.ipeva.fr  -   www.ipeva-studio.com
>
>  *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
>  Le 01/05/2010 ? 21:12, Madovsky a ?crit :
>
>  Hi,
>
> I'd like to set a var (trunk ID)  in the bridge that I can get from my SIP
> phone when the call is accepted.
> channel variables in set / export ?
>
> Thanks
>
> Franck
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From infos at madovsky.org  Sat May  1 12:39:42 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 1 May 2010 15:39:42 -0400
Subject: [Freeswitch-users] get a special var from accepted call
References: <DB9E87BCD48242F9A6C5BE41A912D970@MOBILEE1705>
	<y2l4468a6771005011234s20ce9d39hc286b5b74b5eb281@mail.gmail.com>
Message-ID: <2961A3B22B354D85A0366CCA90084ED1@MOBILEE1705>

Thank you 

F
  ----- Original Message ----- 
  From: Ognjen Seslija 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Saturday, May 01, 2010 3:34 PM
  Subject: Re: [Freeswitch-users] get a special var from accepted call


  <action application="set" data="sip_rh_Foo"/>

  will set Foo on 200 OK received time.

  O.


  On Sat, May 1, 2010 at 9:12 PM, Madovsky <infos at madovsky.org> wrote:

    Hi,

    I'd like to set a var (trunk ID)  in the bridge that I can get from my SIP phone when the call is accepted.
    channel variables in set / export ?

    Thanks

    Franck

    _______________________________________________
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    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From infos at madovsky.org  Sat May  1 12:46:10 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 1 May 2010 15:46:10 -0400
Subject: [Freeswitch-users] get a special var from accepted call
References: <DB9E87BCD48242F9A6C5BE41A912D970@MOBILEE1705><A57E04F6-D0D1-4129-9174-96C98752B655@gmail.com><4BAB40A76D8B4748A3A562FC8A37009F@MOBILEE1705>
	<s2w4468a6771005011239o26875bceza0397e708cab444b@mail.gmail.com>
Message-ID: <2387EC630C90471DAB04C7BA3C726F04@MOBILEE1705>

in fact I'd like to get a var that inform the caller phone which trunk it uses once the call is accepted.
like trunkUsed=bee 

  ----- Original Message ----- 
  From: Ognjen Seslija 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Saturday, May 01, 2010 3:39 PM
  Subject: Re: [Freeswitch-users] get a special var from accepted call


  You need to send a certain SIP header/value pair?



  On Sat, May 1, 2010 at 9:32 PM, Madovsky <infos at madovsky.org> wrote:

    Hi David,

    no, set a var in the dialplan that the caller phone will receive (I use mod_distributor to load balance trunks)

    Thanks
      ----- Original Message ----- 
      From: David Ponzone 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Saturday, May 01, 2010 3:24 PM
      Subject: Re: [Freeswitch-users] get a special var from accepted call


      Franck,


      sorry, can you rephrase that ? 


      You want to set a variable in the dialplan that the called phone will receive ?


      export is your friend then.


      David Ponzone  Direction Technique
      email: david.ponzone at ipeva.fr
      tel:      01 74 03 18 97
      gsm:   06 66 98 76 34


      Service Client IPeva
      tel:      0811 46 26 26
      www.ipeva.fr  -   www.ipeva-studio.com


      Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.








      Le 01/05/2010 ? 21:12, Madovsky a ?crit :


        Hi,

        I'd like to set a var (trunk ID)  in the bridge that I can get from my SIP phone when the call is accepted.
        channel variables in set / export ?

        Thanks

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





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


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      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From david.ponzone at gmail.com  Sat May  1 12:49:41 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Sat, 1 May 2010 21:49:41 +0200
Subject: [Freeswitch-users] get a special var from accepted call
In-Reply-To: <2961A3B22B354D85A0366CCA90084ED1@MOBILEE1705>
References: <DB9E87BCD48242F9A6C5BE41A912D970@MOBILEE1705>
	<y2l4468a6771005011234s20ce9d39hc286b5b74b5eb281@mail.gmail.com>
	<2961A3B22B354D85A0366CCA90084ED1@MOBILEE1705>
Message-ID: <9FBACF75-CAD0-437A-B7AF-9B9490B5530C@gmail.com>

Franck,

you should know that you can also get support through IRC: #freeswitch  
on irc.freenode.net.

Most of the time, it's faster for such quesions.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 01/05/2010 ? 21:39, Madovsky a ?crit :

> Thank you
>
> F
> ----- Original Message -----
> From: Ognjen Seslija
> To: freeswitch-users at lists.freeswitch.org
> Sent: Saturday, May 01, 2010 3:34 PM
> Subject: Re: [Freeswitch-users] get a special var from accepted call
>
> <action application="set" data="sip_rh_Foo"/>
>
> will set Foo on 200 OK received time.
>
> O.
>
> On Sat, May 1, 2010 at 9:12 PM, Madovsky <infos at madovsky.org> wrote:
> Hi,
>
> I'd like to set a var (trunk ID)  in the bridge that I can get from  
> my SIP phone when the call is accepted.
> channel variables in set / export ?
>
> Thanks
>
> Franck
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From infos at madovsky.org  Sat May  1 13:05:05 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 1 May 2010 16:05:05 -0400
Subject: [Freeswitch-users] get a special var from accepted call
References: <DB9E87BCD48242F9A6C5BE41A912D970@MOBILEE1705><y2l4468a6771005011234s20ce9d39hc286b5b74b5eb281@mail.gmail.com><2961A3B22B354D85A0366CCA90084ED1@MOBILEE1705>
	<9FBACF75-CAD0-437A-B7AF-9B9490B5530C@gmail.com>
Message-ID: <040CB4FCD88F48279333930FBF45E768@MOBILEE1705>

ok thanks I will go next time
  ----- Original Message ----- 
  From: David Ponzone 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Saturday, May 01, 2010 3:49 PM
  Subject: Re: [Freeswitch-users] get a special var from accepted call


  Franck,


  you should know that you can also get support through IRC: #freeswitch on irc.freenode.net.


  Most of the time, it's faster for such quesions.


  David Ponzone  Direction Technique
  email: david.ponzone at ipeva.fr
  tel:      01 74 03 18 97
  gsm:   06 66 98 76 34


  Service Client IPeva
  tel:      0811 46 26 26
  www.ipeva.fr  -   www.ipeva-studio.com


  Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.








  Le 01/05/2010 ? 21:39, Madovsky a ?crit :


    Thank you

    F
      ----- Original Message -----
      From: Ognjen Seslija
      To: freeswitch-users at lists.freeswitch.org
      Sent: Saturday, May 01, 2010 3:34 PM
      Subject: Re: [Freeswitch-users] get a special var from accepted call


      <action application="set" data="sip_rh_Foo"/>

      will set Foo on 200 OK received time.

      O.


      On Sat, May 1, 2010 at 9:12 PM, Madovsky <infos at madovsky.org> wrote:

        Hi,

        I'd like to set a var (trunk ID)  in the bridge that I can get from my SIP phone when the call is accepted.
        channel variables in set / export ?

        Thanks

        Franck

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







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



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

    _______________________________________________
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    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From vfclists at googlemail.com  Sat May  1 14:00:03 2010
From: vfclists at googlemail.com (Frank Church)
Date: Sat, 1 May 2010 22:00:03 +0100
Subject: [Freeswitch-users] mod_xml_cdr is not logging at all.
In-Reply-To: <n2z7abab2411005010051ye18ea1f2s5e5cf2e57158d09c@mail.gmail.com>
References: <n2z7abab2411005010051ye18ea1f2s5e5cf2e57158d09c@mail.gmail.com>
Message-ID: <h2v7abab2411005011400u3dcab078zba3e50d5fc6b3418@mail.gmail.com>

Any relief with this question?
It is beginning to feel rather lonely.

On 1 May 2010 08:51, Frank Church <vfclists at googlemail.com> wrote:

> Mod_xml_cdr is not logging anything at all. I have created an additional
> directory called logs, and it is still not working.
> cdr_csv is logging fine.
>
> Below is the current file, are there any faults in it?
> Which directory does {prefix} point to?
>
> My current verson is 17408m on Windows
>
> <configuration name="xml_cdr.conf" description="XML CDR CURL logger">
>>   <settings>
>>     <param name="url" value="http://192.168.1.20:8101/"/>
>>     <param name="log-dir" value="log/xml_cdr"/>
>>     <param name="log-b-leg" value="false"/>
>>     <param name="prefix-a-leg" value="true"/>
>>     <param name="encode" value="true"/>
>>      <param name="err-log-dir" value="log/xml_cdr"/> -->
>>   </settings>
>> </configuration>
>>
>
>
> --
> Frank Church
>
> =======================
> http://devblog.brahmancreations.com
>



-- 
Frank Church

=======================
http://devblog.brahmancreations.com
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From brian at freeswitch.org  Sat May  1 14:17:14 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 1 May 2010 16:17:14 -0500
Subject: [Freeswitch-users] mod_xml_cdr is not logging at all.
In-Reply-To: <h2v7abab2411005011400u3dcab078zba3e50d5fc6b3418@mail.gmail.com>
References: <n2z7abab2411005010051ye18ea1f2s5e5cf2e57158d09c@mail.gmail.com>
	<h2v7abab2411005011400u3dcab078zba3e50d5fc6b3418@mail.gmail.com>
Message-ID: <533EC14F-1C19-4257-81F7-22129FBE2757@freeswitch.org>

That feeling is only going to amplify if you post at 2:54 AM on a Saturday then follow up at 4:03PM on the same Saturday asking about the same question.

Have you done any basic trouble shooting? Make sure mod_xml_cdr is loaded?  Used tcpdump to see if anything is going out on the wire? 

We can't really help without proper information to help diagnose the problem.

/b


On May 1, 2010, at 4:00 PM, Frank Church wrote:

> Any relief with this question?
> It is beginning to feel rather lonely.



From freeswitch.org at todandlorna.com  Sat May  1 14:29:16 2010
From: freeswitch.org at todandlorna.com (Tod Hansmann)
Date: Sat, 01 May 2010 15:29:16 -0600
Subject: [Freeswitch-users] mod_xml_cdr is not logging at all.
In-Reply-To: <h2v7abab2411005011400u3dcab078zba3e50d5fc6b3418@mail.gmail.com>
References: <n2z7abab2411005010051ye18ea1f2s5e5cf2e57158d09c@mail.gmail.com>
	<h2v7abab2411005011400u3dcab078zba3e50d5fc6b3418@mail.gmail.com>
Message-ID: <4BDC9D2C.9090507@todandlorna.com>

Directory permissions maybe?  *shrug*  Could use some more information 
on what steps you've taken. =c)

-Tod Hansmann

On 5/1/2010 3:00 PM, Frank Church wrote:
> Any relief with this question?
> It is beginning to feel rather lonely.
>
> On 1 May 2010 08:51, Frank Church <vfclists at googlemail.com 
> <mailto:vfclists at googlemail.com>> wrote:
>
>     Mod_xml_cdr is not logging anything at all. I have created an
>     additional directory called logs, and it is still not working.
>     cdr_csv is logging fine.
>
>     Below is the current file, are there any faults in it?
>     Which directory does {prefix} point to?
>
>     My current verson is 17408m on Windows
>
>         <configuration name="xml_cdr.conf" description="XML CDR CURL
>         logger">
>         <settings>
>         <param name="url" value="http://192.168.1.20:8101/"/>
>         <param name="log-dir" value="log/xml_cdr"/>
>         <param name="log-b-leg" value="false"/>
>         <param name="prefix-a-leg" value="true"/>
>         <param name="encode" value="true"/>
>         <param name="err-log-dir" value="log/xml_cdr"/> -->
>         </settings>
>         </configuration>
>
>
>
>     -- 
>     Frank Church
>
>     =======================
>     http://devblog.brahmancreations.com
>
>
>
>
> -- 
> Frank Church
>
> =======================
> http://devblog.brahmancreations.com
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    
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From vfclists at googlemail.com  Sat May  1 15:11:22 2010
From: vfclists at googlemail.com (Frank Church)
Date: Sat, 1 May 2010 23:11:22 +0100
Subject: [Freeswitch-users] mod_xml_cdr is not logging at all.
In-Reply-To: <4BDC9D2C.9090507@todandlorna.com>
References: <n2z7abab2411005010051ye18ea1f2s5e5cf2e57158d09c@mail.gmail.com>
	<h2v7abab2411005011400u3dcab078zba3e50d5fc6b3418@mail.gmail.com>
	<4BDC9D2C.9090507@todandlorna.com>
Message-ID: <o2t7abab2411005011511ha22b18dfs1b599c3c74de8e87@mail.gmail.com>

mox_xml_cdr is working fine. I am sending the XML to webserver and saving
the information into a database. It is the file logging which is not working
at all.

What I don't know is if the file path I am using is correct.

cdr_csv works fine and it is a Windows system I don't think it is a file
permisions problem.

On 1 May 2010 22:29, Tod Hansmann <freeswitch.org at todandlorna.com> wrote:

>  Directory permissions maybe?  *shrug*  Could use some more information on
> what steps you've taken. =c)
>
> -Tod Hansmann
>
>
> On 5/1/2010 3:00 PM, Frank Church wrote:
>
> Any relief with this question?
> It is beginning to feel rather lonely.
>
> On 1 May 2010 08:51, Frank Church <vfclists at googlemail.com> wrote:
>
>> Mod_xml_cdr is not logging anything at all. I have created an additional
>> directory called logs, and it is still not working.
>> cdr_csv is logging fine.
>>
>> Below is the current file, are there any faults in it?
>> Which directory does {prefix} point to?
>>
>> My current verson is 17408m on Windows
>>
>> <configuration name="xml_cdr.conf" description="XML CDR CURL logger">
>>>   <settings>
>>>     <param name="url" value="http://192.168.1.20:8101/"/>
>>>     <param name="log-dir" value="log/xml_cdr"/>
>>>     <param name="log-b-leg" value="false"/>
>>>     <param name="prefix-a-leg" value="true"/>
>>>     <param name="encode" value="true"/>
>>>      <param name="err-log-dir" value="log/xml_cdr"/> -->
>>>   </settings>
>>> </configuration>
>>>
>>
>>
>> --
>> Frank Church
>>
>> =======================
>> http://devblog.brahmancreations.com
>>
>
>
>
> --
> Frank Church
>
> =======================
> http://devblog.brahmancreations.com
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Frank Church

=======================
http://devblog.brahmancreations.com
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From vfclists at googlemail.com  Sat May  1 15:18:25 2010
From: vfclists at googlemail.com (Frank Church)
Date: Sat, 1 May 2010 23:18:25 +0100
Subject: [Freeswitch-users] mod_xml_cdr is not logging at all.
In-Reply-To: <533EC14F-1C19-4257-81F7-22129FBE2757@freeswitch.org>
References: <n2z7abab2411005010051ye18ea1f2s5e5cf2e57158d09c@mail.gmail.com>
	<h2v7abab2411005011400u3dcab078zba3e50d5fc6b3418@mail.gmail.com>
	<533EC14F-1C19-4257-81F7-22129FBE2757@freeswitch.org>
Message-ID: <l2n7abab2411005011518t6b03b9edvb995b58df1a89664@mail.gmail.com>

mod_xml_cdr is loaded and sending xml via http fine. Logging is not working.

I have sussed out that the leading Freeswitch developers are in the Western
part of USA and I believe you are 6 - 8 hrs behind London, which is why my
posting times appear odd.

You once remarked that it was late when you answered by question and that
was a time a had just risen from bed.


On 1 May 2010 22:17, Brian West <brian at freeswitch.org> wrote:

> That feeling is only going to amplify if you post at 2:54 AM on a Saturday
> then follow up at 4:03PM on the same Saturday asking about the same
> question.
>
> Have you done any basic trouble shooting? Make sure mod_xml_cdr is loaded?
>  Used tcpdump to see if anything is going out on the wire?
>
> We can't really help without proper information to help diagnose the
> problem.
>
> /b
>
>
> On May 1, 2010, at 4:00 PM, Frank Church wrote:
>
> > Any relief with this question?
> > It is beginning to feel rather lonely.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Frank Church

=======================
http://devblog.brahmancreations.com
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From brian at freeswitch.org  Sat May  1 15:22:32 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 1 May 2010 17:22:32 -0500
Subject: [Freeswitch-users] mod_xml_cdr is not logging at all.
In-Reply-To: <o2t7abab2411005011511ha22b18dfs1b599c3c74de8e87@mail.gmail.com>
References: <n2z7abab2411005010051ye18ea1f2s5e5cf2e57158d09c@mail.gmail.com>
	<h2v7abab2411005011400u3dcab078zba3e50d5fc6b3418@mail.gmail.com>
	<4BDC9D2C.9090507@todandlorna.com>
	<o2t7abab2411005011511ha22b18dfs1b599c3c74de8e87@mail.gmail.com>
Message-ID: <33F909CA-9B32-465A-8090-B3A6DA245821@freeswitch.org>

Well I opened up mod_xml_cdr.c and looked for the 'open' call which brings me to this option in your config file you'll need to set "log-http-and-disk"

<param name="log-http-and-disk" value="true"/>

I have updated the config samples to include this.

/b


On May 1, 2010, at 5:11 PM, Frank Church wrote:

> mox_xml_cdr is working fine. I am sending the XML to webserver and saving the information into a database. It is the file logging which is not working at all.
> 
> What I don't know is if the file path I am using is correct.
> 
> cdr_csv works fine and it is a Windows system I don't think it is a file permisions problem.



From david.ponzone at gmail.com  Sat May  1 15:26:28 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Sun, 2 May 2010 00:26:28 +0200
Subject: [Freeswitch-users] mod_xml_cdr is not logging at all.
In-Reply-To: <o2t7abab2411005011511ha22b18dfs1b599c3c74de8e87@mail.gmail.com>
References: <n2z7abab2411005010051ye18ea1f2s5e5cf2e57158d09c@mail.gmail.com>
	<h2v7abab2411005011400u3dcab078zba3e50d5fc6b3418@mail.gmail.com>
	<4BDC9D2C.9090507@todandlorna.com>
	<o2t7abab2411005011511ha22b18dfs1b599c3c74de8e87@mail.gmail.com>
Message-ID: <EB5AE6C1-8718-4190-8745-319631D65D99@gmail.com>

Frank,

In my config, I have:

<param name="log-dir" value=""/>

This should be enough to get the XML CDRs written to $freeswitch_dir/ 
log/xml_cdr/

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 02/05/2010 ? 00:11, Frank Church a ?crit :

> mox_xml_cdr is working fine. I am sending the XML to webserver and  
> saving the information into a database. It is the file logging which  
> is not working at all.
>
> What I don't know is if the file path I am using is correct.
>
> cdr_csv works fine and it is a Windows system I don't think it is a  
> file permisions problem.
>
> On 1 May 2010 22:29, Tod Hansmann <freeswitch.org at todandlorna.com>  
> wrote:
> Directory permissions maybe?  *shrug*  Could use some more  
> information on what steps you've taken. =c)
>
> -Tod Hansmann
>
>
> On 5/1/2010 3:00 PM, Frank Church wrote:
>> Any relief with this question?
>> It is beginning to feel rather lonely.
>>
>> On 1 May 2010 08:51, Frank Church <vfclists at googlemail.com> wrote:
>> Mod_xml_cdr is not logging anything at all. I have created an  
>> additional directory called logs, and it is still not working.
>> cdr_csv is logging fine.
>>
>> Below is the current file, are there any faults in it?
>> Which directory does {prefix} point to?
>>
>> My current verson is 17408m on Windows
>>
>> <configuration name="xml_cdr.conf" description="XML CDR CURL logger">
>>   <settings>
>>     <param name="url" value="http://192.168.1.20:8101/"/>
>>     <param name="log-dir" value="log/xml_cdr"/>
>>     <param name="log-b-leg" value="false"/>
>>     <param name="prefix-a-leg" value="true"/>
>>     <param name="encode" value="true"/>
>>      <param name="err-log-dir" value="log/xml_cdr"/> -->
>>   </settings>
>> </configuration>
>>
>>
>> -- 
>> Frank Church
>>
>> =======================
>> http://devblog.brahmancreations.com
>>
>>
>>
>> -- 
>> Frank Church
>>
>> =======================
>> http://devblog.brahmancreations.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Frank Church
>
> =======================
> http://devblog.brahmancreations.com
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From nico at vthadden.de  Sat May  1 15:29:29 2010
From: nico at vthadden.de (Nicola von Thadden)
Date: Sun, 02 May 2010 00:29:29 +0200
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
 freeswitch (ptime:30 msecs)
In-Reply-To: <ADF49E76-C340-448A-B4F9-4B186E39660B@freeswitch.org>
References: <4BD86880.4090607@vthadden.de>	<h2m191c3a031004281404v4b9a5cdcp37cf1f4f621222a7@mail.gmail.com>	<4BD8A5CC.8060107@vthadden.de>	<s2k191c3a031004281426za5fd5755r13d3e460d227c1d5@mail.gmail.com>	<4BD98D57.7030100@gmx.net>	<1AC6CE8D-A1CF-4128-AB7F-027CD35B0F08@avgs.ca>	<4BD9952E.604@gmx.net>	<1A371C4C-1225-45C9-AFAB-999B53D52AF6@avgs.ca>	<t2i191c3a031004290801qc46319a8w153186d0939e34aa@mail.gmail.com>	<z2wc96b95651004290834t39ddb87mbb61a8c9cf37b036@mail.gmail.com>	<2921DCA5-BF5F-4345-8011-7F00D2AA5D6A@freeswitch.org>	<4BD9AE67.2040600@gmx.net>
	<ADF49E76-C340-448A-B4F9-4B186E39660B@freeswitch.org>
Message-ID: <4BDCAB49.8070200@vthadden.de>

It's not so easy to find something with the same or comparable features:
ADSL2+ (16mbit), ISDN and normal landline, VOIP, DECT, WLAN (a/b/g/n), etc

Additionally they are sponsored by many ISPs, so you get them for free
(or with a lot of discount).

I have just recompiled from latest git version (done with the Makefile
given here: http://www.freeswitch.org/eg/Makefile) and have the same
issues as always.
Setting rtp-autofix-timing to false removed the choppiness but added an
delay in conferences (5 to 10 seconds)...
CPU is between 40 and 170% (on a quadcore) and ram is just 20mb used by
freeswitch, so this shouldn't be the problem.

Nico

On 29.04.2010 18:13, Brian West wrote:
> But you have exactly three of them.  Its cheaper to go buy something thats not broken vs trying to work around this.
> 
> Just saying!
> 
> /b
> 
> On Apr 29, 2010, at 11:05 AM, Peter P GMX wrote:
> 
>> I have 3 of them, in Germany I think there are about 10 Mio. Plus
>> France, Austria,Switzerland etc. I would say some dozens of Millions.
>>
>> Best regards
>> Peter
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From anthony.minessale at gmail.com  Sat May  1 15:55:45 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 1 May 2010 17:55:45 -0500
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
	freeswitch (ptime:30 msecs)
In-Reply-To: <4BDCAB49.8070200@vthadden.de>
References: <4BD86880.4090607@vthadden.de>
	<1AC6CE8D-A1CF-4128-AB7F-027CD35B0F08@avgs.ca> <4BD9952E.604@gmx.net>
	<1A371C4C-1225-45C9-AFAB-999B53D52AF6@avgs.ca>
	<t2i191c3a031004290801qc46319a8w153186d0939e34aa@mail.gmail.com>
	<z2wc96b95651004290834t39ddb87mbb61a8c9cf37b036@mail.gmail.com>
	<2921DCA5-BF5F-4345-8011-7F00D2AA5D6A@freeswitch.org>
	<4BD9AE67.2040600@gmx.net>
	<ADF49E76-C340-448A-B4F9-4B186E39660B@freeswitch.org>
	<4BDCAB49.8070200@vthadden.de>
Message-ID: <u2w191c3a031005011555v31733ef4if3b9f41b2536b9be@mail.gmail.com>

every time you repost saying it's still not working I will reply telling you
that its because the fritzbox is broken.  I can assure you some combination
or another of the advice we gave you probably can work around the problem
and you missed it.  In short, we will never have some magic patch to make it
start working right because it's the other side who is broken not us.

Maybe you should find another user who owns one and managed to hack it into
working.


On Sat, May 1, 2010 at 5:29 PM, Nicola von Thadden <nico at vthadden.de> wrote:

> It's not so easy to find something with the same or comparable features:
> ADSL2+ (16mbit), ISDN and normal landline, VOIP, DECT, WLAN (a/b/g/n), etc
>
> Additionally they are sponsored by many ISPs, so you get them for free
> (or with a lot of discount).
>
> I have just recompiled from latest git version (done with the Makefile
> given here: http://www.freeswitch.org/eg/Makefile) and have the same
> issues as always.
> Setting rtp-autofix-timing to false removed the choppiness but added an
> delay in conferences (5 to 10 seconds)...
> CPU is between 40 and 170% (on a quadcore) and ram is just 20mb used by
> freeswitch, so this shouldn't be the problem.
>
> Nico
>
> On 29.04.2010 18:13, Brian West wrote:
> > But you have exactly three of them.  Its cheaper to go buy something
> thats not broken vs trying to work around this.
> >
> > Just saying!
> >
> > /b
> >
> > On Apr 29, 2010, at 11:05 AM, Peter P GMX wrote:
> >
> >> I have 3 of them, in Germany I think there are about 10 Mio. Plus
> >> France, Austria,Switzerland etc. I would say some dozens of Millions.
> >>
> >> Best regards
> >> Peter
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <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
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From david.ponzone at gmail.com  Sat May  1 16:21:06 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Sun, 2 May 2010 01:21:06 +0200
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
	freeswitch (ptime:30 msecs)
In-Reply-To: <4BD9AE67.2040600@gmx.net>
References: <4BD86880.4090607@vthadden.de>	<h2m191c3a031004281404v4b9a5cdcp37cf1f4f621222a7@mail.gmail.com>	<4BD8A5CC.8060107@vthadden.de>	<s2k191c3a031004281426za5fd5755r13d3e460d227c1d5@mail.gmail.com>	<4BD98D57.7030100@gmx.net>	<1AC6CE8D-A1CF-4128-AB7F-027CD35B0F08@avgs.ca>	<4BD9952E.604@gmx.net>	<1A371C4C-1225-45C9-AFAB-999B53D52AF6@avgs.ca>	<t2i191c3a031004290801qc46319a8w153186d0939e34aa@mail.gmail.com>	<z2wc96b95651004290834t39ddb87mbb61a8c9cf37b036@mail.gmail.com>
	<2921DCA5-BF5F-4345-8011-7F00D2AA5D6A@freeswitch.org>
	<4BD9AE67.2040600@gmx.net>
Message-ID: <60F5ACFC-9F5F-4E72-A4E3-FE98F6BDCB2F@gmail.com>

Peter,

just for the record, this box is absolutely unknown in France.
The french market is saturated with triple-play boxes as they are  
provided for free by ISPs and the leading ISPs probably have around  
95% of the market.

I would not speak for the other countries, though.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 29/04/2010 ? 18:05, Peter P GMX a ?crit :

> I have 3 of them, in Germany I think there are about 10 Mio. Plus
> France, Austria,Switzerland etc. I would say some dozens of Millions.
>
> Best regards
> Peter
>
> Brian West schrieb:
>> OK how many of these devices are we talking about that you have?
>>
>> /b
>>
>> On Apr 29, 2010, at 10:34 AM, Saeed Ahmed wrote:
>>
>>> i think usually not, because with FS sometimes you get the patch
>>> within minutes, which you'll never get somewhere else :-)
>>>
>>> On Thu, Apr 29, 2010 at 5:01 PM, Anthony
>>> Minessale <anthony.minessale at gmail.com
>>> <mailto:anthony.minessale at gmail.com>> wrote:
>>>
>>>    Does anyone ever report this issue to Fritzbox or is too hard
>>>    to conceive that they have the bug in this situation?
>>>
>>
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Sat May  1 16:25:13 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 1 May 2010 18:25:13 -0500
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
	freeswitch (ptime:30 msecs)
In-Reply-To: <60F5ACFC-9F5F-4E72-A4E3-FE98F6BDCB2F@gmail.com>
References: <4BD86880.4090607@vthadden.de>	<h2m191c3a031004281404v4b9a5cdcp37cf1f4f621222a7@mail.gmail.com>	<4BD8A5CC.8060107@vthadden.de>	<s2k191c3a031004281426za5fd5755r13d3e460d227c1d5@mail.gmail.com>	<4BD98D57.7030100@gmx.net>	<1AC6CE8D-A1CF-4128-AB7F-027CD35B0F08@avgs.ca>	<4BD9952E.604@gmx.net>	<1A371C4C-1225-45C9-AFAB-999B53D52AF6@avgs.ca>	<t2i191c3a031004290801qc46319a8w153186d0939e34aa@mail.gmail.com>	<z2wc96b95651004290834t39ddb87mbb61a8c9cf37b036@mail.gmail.com>
	<2921DCA5-BF5F-4345-8011-7F00D2AA5D6A@freeswitch.org>
	<4BD9AE67.2040600@gmx.net>
	<60F5ACFC-9F5F-4E72-A4E3-FE98F6BDCB2F@gmail.com>
Message-ID: <852299E9-A5B9-4047-B8CD-053A1936429A@freeswitch.org>

Can I ship you a hammer already?

/b

On May 1, 2010, at 6:21 PM, David Ponzone wrote:

> Peter,
> 
> just for the record, this box is absolutely unknown in France.
> The french market is saturated with triple-play boxes as they are provided for free by ISPs and the leading ISPs probably have around 95% of the market.
> 
> I would not speak for the other countries, though.



From sean at obscuradigital.com  Sat May  1 18:30:37 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Sat, 01 May 2010 18:30:37 -0700
Subject: [Freeswitch-users] Connect external phone
Message-ID: <C80223CD.935B%sean@obscuradigital.com>

Hello list,

I keep getting this authentication error when I attempt to connect a device
from the outside.  I followed the wiki that suggest creating a separate
profile to handle external device connections.

Here?s my external xml file
<profile name="gexternal5090">
  <!-- This profile is only for outbound registrations to providers -->
  <gateways>
    <X-PRE-PROCESS cmd="include" data="external/*.xml"/>
  </gateways>
  <aliases>
    <alias name="outbound"/>
  </aliases>
  <domains>
    <domain name="$${domain}" parse="true"/>
  </domains>
  <settings>
    <param name="debug" value="0"/>
    <param name="sip-trace" value="no"/>
    <param name="rfc2833-pt" value="101"/>
    <param name="sip-port" value="5090"/>
    <param name="dialplan" value="XML"/>
    <param name="context" value="public"/>
    <param name="dtmf-duration" value="100"/>
    <param name="codec-prefs" value="$${outbound_codec_prefs}"/>
    <param name="hold-music" value="$${moh_uri}"/>
    <param name="use-rtp-timer" value="true"/>
    <param name="rtp-timer-name" value="soft"/>
    <param name="manage-presence" value="false"/>
    <param name="inbound-codec-negotiation" value="generous"/>
    <param name="nonce-ttl" value="60"/>
    <param name="auth-calls" value="false"/>
    <param name="rtp-timeout-sec" value="1800"/>
    <param name="rtp-ip" value="$${local_ip_v4}"/>
    <param name="sip-ip" value="$${local_ip_v4}"/>
    <param name="ext-rtp-ip" value="71.133.39.219"/>
    <param name="ext-sip-ip" value="71.133.39.219"/>
    <param name="rtp-timeout-sec" value="300"/>
    <param name="rtp-hold-timeout-sec" value="1800"/>
  </settings>
</profile>

I did a siptrace on the incoming connection attempt
  ------------------------------------------------------------------------
   REGISTER sip:71.133.39.219:5090 SIP/2.0
   Via: SIP/2.0/UDP x.x.x.x:5090;branch=z9hG4bKvvaun9r6.1u8ej
   Max-Forwards: 70
   From: <sip:1001 at 71.133.39.219>;tag=i9w3pt4bm4tpsl
   To: <sip:1001 at 71.133.39.219>
   Call-ID: fo33ia7sk6qy9vnk6h
   CSeq: 75203 REGISTER
   Contact: <sip:1001 at x.x.x.x:5090;line=20880>
   Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE,
NOTIFY, MESSAGE, INFO, PRACK
   Expires: 600
   User-Agent: snom-m3-SIP/02.02 (MAC=0004132AD0BE; HW=255)
   Content-Length: 0
   
   ------------------------------------------------------------------------
send 591 bytes to udp/[x.x.x.x]:5090 at 01:26:47.109165:
   ------------------------------------------------------------------------
   SIP/2.0 401 Unauthorized
   Via: SIP/2.0/UDP
10.0.10.231:5090;branch=z9hG4bKvvaun9r6.1u8ej;received=207.47.31.108
   From: <sip:1001 at 71.133.39.219>;tag=i9w3pt4bm4tpsl
   To: <sip:1001 at 71.133.39.219>;tag=Zc7mBaBN723Bp
   Call-ID: fo33ia7sk6qy9vnk6h
   CSeq: 75203 REGISTER
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   WWW-Authenticate: Digest realm="71.133.39.219",
nonce="bd1ecf94-5589-11df-b3f4-c1a375adf51a", algorithm=MD5, qop="auth"
   Content-Length: 0
   
   ------------------------------------------------------------------------
 
Thoughts?
Thanks
Sean
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From david.ponzone at gmail.com  Sun May  2 01:00:19 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Sun, 2 May 2010 10:00:19 +0200
Subject: [Freeswitch-users] Connect external phone
In-Reply-To: <C80223CD.935B%sean@obscuradigital.com>
References: <C80223CD.935B%sean@obscuradigital.com>
Message-ID: <428AE2E1-1036-4331-B0B3-A685EAABB9B0@gmail.com>

Sean,

my suggestion would be not to reinvent the wheel.
If you need another profile to accept registrations, just copy  
internal.xml and modify what you need.

But I think there is another issue here.
FS sends back a 401, which is normal, but your SNOM should then send  
another REGISTER with Digest auth.
It doesn't OR your failed to include the whole trace.
If it doesn't, it means it's screwed.
I had this issue with a very bad ATA from Telco Systems.
But I think some people use the M3 on FS, so I guess there is way to  
make that work (perhaps a firmware update on the M3 or a parameter).

First thing, can you confirm ASAP you included the whole trace ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 02/05/2010 ? 03:30, Sean Holt a ?crit :

> Hello list,
>
> I keep getting this authentication error when I attempt to connect a  
> device from the outside.  I followed the wiki that suggest creating  
> a separate profile to handle external device connections.
>
> Here?s my external xml file
> <profile name="gexternal5090">
>   <!-- This profile is only for outbound registrations to providers  
> -->
>   <gateways>
>     <X-PRE-PROCESS cmd="include" data="external/*.xml"/>
>   </gateways>
>   <aliases>
>     <alias name="outbound"/>
>   </aliases>
>   <domains>
>     <domain name="$${domain}" parse="true"/>
>   </domains>
>   <settings>
>     <param name="debug" value="0"/>
>     <param name="sip-trace" value="no"/>
>     <param name="rfc2833-pt" value="101"/>
>     <param name="sip-port" value="5090"/>
>     <param name="dialplan" value="XML"/>
>     <param name="context" value="public"/>
>     <param name="dtmf-duration" value="100"/>
>     <param name="codec-prefs" value="$${outbound_codec_prefs}"/>
>     <param name="hold-music" value="$${moh_uri}"/>
>     <param name="use-rtp-timer" value="true"/>
>     <param name="rtp-timer-name" value="soft"/>
>     <param name="manage-presence" value="false"/>
>     <param name="inbound-codec-negotiation" value="generous"/>
>     <param name="nonce-ttl" value="60"/>
>     <param name="auth-calls" value="false"/>
>     <param name="rtp-timeout-sec" value="1800"/>
>     <param name="rtp-ip" value="$${local_ip_v4}"/>
>     <param name="sip-ip" value="$${local_ip_v4}"/>
>     <param name="ext-rtp-ip" value="71.133.39.219"/>
>     <param name="ext-sip-ip" value="71.133.39.219"/>
>     <param name="rtp-timeout-sec" value="300"/>
>     <param name="rtp-hold-timeout-sec" value="1800"/>
>   </settings>
> </profile>
>
> I did a siptrace on the incoming connection attempt
>    
> ------------------------------------------------------------------------
>    REGISTER sip:71.133.39.219:5090 SIP/2.0
>    Via: SIP/2.0/UDP x.x.x.x:5090;branch=z9hG4bKvvaun9r6.1u8ej
>    Max-Forwards: 70
>    From: <sip:1001 at 71.133.39.219>;tag=i9w3pt4bm4tpsl
>    To: <sip:1001 at 71.133.39.219>
>    Call-ID: fo33ia7sk6qy9vnk6h
>    CSeq: 75203 REGISTER
>    Contact: <sip:1001 at x.x.x.x:5090;line=20880>
>    Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER,  
> SUBSCRIBE, NOTIFY, MESSAGE, INFO, PRACK
>    Expires: 600
>    User-Agent: snom-m3-SIP/02.02 (MAC=0004132AD0BE; HW=255)
>    Content-Length: 0
>
>     
> ------------------------------------------------------------------------
> send 591 bytes to udp/[x.x.x.x]:5090 at 01:26:47.109165:
>     
> ------------------------------------------------------------------------
>    SIP/2.0 401 Unauthorized
>    Via: SIP/2.0/UDP  
> 10.0.10.231:5090;branch=z9hG4bKvvaun9r6.1u8ej;received=207.47.31.108
>    From: <sip:1001 at 71.133.39.219>;tag=i9w3pt4bm4tpsl
>    To: <sip:1001 at 71.133.39.219>;tag=Zc7mBaBN723Bp
>    Call-ID: fo33ia7sk6qy9vnk6h
>    CSeq: 75203 REGISTER
>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,  
> REGISTER, REFER, NOTIFY
>    Supported: timer, precondition, path, replaces
>    WWW-Authenticate: Digest realm="71.133.39.219",  
> nonce="bd1ecf94-5589-11df-b3f4-c1a375adf51a", algorithm=MD5,  
> qop="auth"
>    Content-Length: 0
>
>     
> ------------------------------------------------------------------------
>
> Thoughts?
> Thanks
> Sean
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From babak.freeswitch at gmail.com  Sun May  2 01:22:09 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Sun, 2 May 2010 12:52:09 +0430
Subject: [Freeswitch-users] nat problem
Message-ID: <q2g1fdf92721005020122n23fc1f6x89bfa9532123f520@mail.gmail.com>

Hi
I got two 7941 ip phones connected to freeswitch (ip phone)192.168.22.57
-> 192.168.22.1 -> 192.168.11.30(fs)
they can register fine but I can not call from ip phone to ip phone. when I
use 3cx softphone in the same conditions I can call from ip phone to 3cx but
not from 3cx to ip phone this is result of  "sofia status profile internal"
and when I get pcap on fs when invite is sent to ip phones nothing comes
back


Call-ID:        001bd432-a95c0003-942f6b14-17330728 at 192.168.22.56
User:           1005 at 192.168.11.30
Contact:        "user" <sip:1005 at 192.168.22.56:5060
;transport=udp;fs_nat=yes;fs_
path=sip%3A1005%40192.168.22.56%3A49551%3Btransport%3Dudp>
Agent:          Cisco-CP7941G/8.5.2
Status:         Registered(UDP-NAT)(unknown) EXP(2010-05-02 14:11:11)
Host:           s-efrn2brk8zpcy
IP:             192.168.22.56
Port:           49551
Auth-User:      1005
Auth-Realm:     192.168.11.30
MWI-Account:    1005 at 192.168.11.30

Call-ID:        001bd433-4c190002-d08bed92-50fab18f at 192.168.22.57
User:           1010 at 192.168.11.30
Contact:        "user" <sip:1010 at 192.168.22.57:5060
;transport=udp;fs_nat=yes;fs_
path=sip%3A1010%40192.168.22.57%3A51471%3Btransport%3Dudp>
Agent:          Cisco-CP7941G/8.5.2
Status:         Registered(UDP-NAT)(unknown) EXP(2010-05-02 12:46:04)
Host:           s-efrn2brk8zpcy
IP:             192.168.22.57
Port:           51471
Auth-User:      1010
Auth-Realm:     192.168.11.30
MWI-Account:    1010 at 192.168.11.30

Call-ID:        MjYzNmQ5YWYyMDkwMDc3M2UxYzY5Y2JkZGE4NTBhZWU.
User:           1001 at 192.168.11.30
Contact:        "3CXPhone" <sip:1001 at 192.168.22.58:3884
;rinstance=74bf6bb1ef7db1
27;fs_nat=yes;fs_path=sip%3A1001%40192.168.22.58%3A3884%3Brinstance%3D74bf6bb1ef
7db127>
Agent:          3CXVoipPhone 4.0.9878.0
Status:         Registered(UDP-NAT)(unknown) EXP(2010-05-02 12:45:37)
Host:           s-efrn2brk8zpcy
IP:             192.168.22.58
Port:           3884
Auth-User:      1001
Auth-Realm:     192.168.11.30
MWI-Account:    1001 at 192.168.11.30
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From sean at obscuradigital.com  Sun May  2 09:56:53 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Sun, 02 May 2010 09:56:53 -0700
Subject: [Freeswitch-users] Connect external phone
In-Reply-To: <428AE2E1-1036-4331-B0B3-A685EAABB9B0@gmail.com>
Message-ID: <C802FCE5.936B%sean@obscuradigital.com>

David,

I did double check firmware version for my phone, it?s the most up-to-date.
I have been able to connect the phone inside the the local lan no problem,
I?m attempting to test an external connection for office employee?s who are
in different city locations.

I turned on trace using this command, sofia profile gexternal siptrace on.
The only logging that I saw referencing M3 I provided below.  I could turn
on sofia loglevel for additional info if that helps.

I wonder, since my phone is behind a NAT if that?s causing an issue.

Thanks
Sean


On 5/2/10 1:00 AM, "David Ponzone" <david.ponzone at gmail.com> wrote:

> Sean,
> 
> my suggestion would be not to reinvent the wheel.
> If you need another profile to accept registrations, just copy internal.xml
> and modify what you need.
> 
> But I think there is another issue here.
> FS sends back a 401, which is normal, but your SNOM should then send another
> REGISTER with Digest auth.
> It doesn't OR your failed to include the whole trace.
> If it doesn't, it means it's screwed.
> I had this issue with a very bad ATA from Telco Systems.
> But I think some people use the M3 on FS, so I guess there is way to make that
> work (perhaps a firmware update on the M3 or a parameter).
> 
> First thing, can you confirm ASAP you included the whole trace ?
> 
>  
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>   -   www.ipeva-studio.com
> <BLOCKED::http://www.ipeva-studio.com/>
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non
> autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a
> ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
>  
> 
> Le 02/05/2010 ? 03:30, Sean Holt a ?crit :
> 
>>  Hello list,
>>  
>>  I keep getting this authentication error when I attempt to connect a device
>> from the outside.  I followed the wiki that suggest creating a separate
>> profile to handle external device connections.
>>  
>>  Here?s my external xml file
>>  <profile name="gexternal5090">
>>    <!-- This profile is only for outbound registrations to providers -->
>>    <gateways>
>>      <X-PRE-PROCESS cmd="include" data="external/*.xml"/>
>>    </gateways>
>>    <aliases>
>>      <alias name="outbound"/>
>>    </aliases>
>>    <domains>
>>      <domain name="$${domain}" parse="true"/>
>>    </domains>
>>    <settings>
>>      <param name="debug" value="0"/>
>>      <param name="sip-trace" value="no"/>
>>      <param name="rfc2833-pt" value="101"/>
>>      <param name="sip-port" value="5090"/>
>>      <param name="dialplan" value="XML"/>
>>      <param name="context" value="public"/>
>>      <param name="dtmf-duration" value="100"/>
>>      <param name="codec-prefs" value="$${outbound_codec_prefs}"/>
>>      <param name="hold-music" value="$${moh_uri}"/>
>>      <param name="use-rtp-timer" value="true"/>
>>      <param name="rtp-timer-name" value="soft"/>
>>      <param name="manage-presence" value="false"/>
>>      <param name="inbound-codec-negotiation" value="generous"/>
>>      <param name="nonce-ttl" value="60"/>
>>      <param name="auth-calls" value="false"/>
>>      <param name="rtp-timeout-sec" value="1800"/>
>>      <param name="rtp-ip" value="$${local_ip_v4}"/>
>>      <param name="sip-ip" value="$${local_ip_v4}"/>
>>      <param name="ext-rtp-ip" value="71.133.39.219"/>
>>      <param name="ext-sip-ip" value="71.133.39.219"/>
>>      <param name="rtp-timeout-sec" value="300"/>
>>      <param name="rtp-hold-timeout-sec" value="1800"/>
>>    </settings>
>>  </profile>
>>  
>>  I did a siptrace on the incoming connection attempt
>>    ------------------------------------------------------------------------
>>     REGISTER sip:71.133.39.219:5090 SIP/2.0
>>     Via: SIP/2.0/UDP x.x.x.x:5090;branch=z9hG4bKvvaun9r6.1u8ej
>>     Max-Forwards: 70
>>     From: <sip:1001 at 71.133.39.219>;tag=i9w3pt4bm4tpsl
>>     To: <sip:1001 at 71.133.39.219>
>>     Call-ID: fo33ia7sk6qy9vnk6h
>>     CSeq: 75203 REGISTER
>>     Contact: <sip:1001 at x.x.x.x:5090;line=20880>
>>     Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE,
>> NOTIFY, MESSAGE, INFO, PRACK
>>     Expires: 600
>>     User-Agent: snom-m3-SIP/02.02 (MAC=0004132AD0BE; HW=255)
>>     Content-Length: 0
>>     
>>     ------------------------------------------------------------------------
>>  send 591 bytes to udp/[x.x.x.x]:5090 at 01:26:47.109165:
>>     ------------------------------------------------------------------------
>>     SIP/2.0 401 Unauthorized
>>     Via: SIP/2.0/UDP
>> 10.0.10.231:5090;branch=z9hG4bKvvaun9r6.1u8ej;received=207.47.31.108
>>     From: <sip:1001 at 71.133.39.219>;tag=i9w3pt4bm4tpsl
>>     To: <sip:1001 at 71.133.39.219>;tag=Zc7mBaBN723Bp
>>     Call-ID: fo33ia7sk6qy9vnk6h
>>     CSeq: 75203 REGISTER
>>     User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-
>>     Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
>> REGISTER, REFER, NOTIFY
>>     Supported: timer, precondition, path, replaces
>>     WWW-Authenticate: Digest realm="71.133.39.219",
>> nonce="bd1ecf94-5589-11df-b3f4-c1a375adf51a", algorithm=MD5, qop="auth"
>>     Content-Length: 0
>>     
>>     ------------------------------------------------------------------------
>>   
>>  Thoughts?
>>  Thanks
>>  Sean 
>>   _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From msc at freeswitch.org  Sun May  2 10:01:04 2010
From: msc at freeswitch.org (Michael S Collins)
Date: Sun, 2 May 2010 10:01:04 -0700
Subject: [Freeswitch-users] Connect external phone
In-Reply-To: <428AE2E1-1036-4331-B0B3-A685EAABB9B0@gmail.com>
References: <C80223CD.935B%sean@obscuradigital.com>
	<428AE2E1-1036-4331-B0B3-A685EAABB9B0@gmail.com>
Message-ID: <5003722E-10DF-4D16-8DF9-E7B973FBCDE8@freeswitch.org>

It's been my experience that the snoms work okay in a remote scenario.  
I would try using 5060/internal.xml first. Make sure it works there  
and then do what David P suggested about copying internal.xml and  
creating a new profile.

-MC

Sent from my iPhone

On May 2, 2010, at 1:00 AM, David Ponzone <david.ponzone at gmail.com>  
wrote:

> Sean,
>
> my suggestion would be not to reinvent the wheel.
> If you need another profile to accept registrations, just copy  
> internal.xml and modify what you need.
>
> But I think there is another issue here.
> FS sends back a 401, which is normal, but your SNOM should then send  
> another REGISTER with Digest auth.
> It doesn't OR your failed to include the whole trace.
> If it doesn't, it means it's screwed.
> I had this issue with a very bad ATA from Telco Systems.
> But I think some people use the M3 on FS, so I guess there is way to  
> make that work (perhaps a firmware update on the M3 or a parameter).
>
> First thing, can you confirm ASAP you included the whole trace ?
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tabli 
> s ? l'intention exclusive de ses destinataires. Toute utilisation ou 
>  diffusion non autoris?e est interdite. Tout message ?lectronique es 
> t susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
> vous n'?tes pas destinataire de ce message, merci de le d?truire imm 
> ?diatement et d'avertir l'exp?diteur.
>
>
>
>
> Le 02/05/2010 ? 03:30, Sean Holt a ?crit :
>
>> Hello list,
>>
>> I keep getting this authentication error when I attempt to connect  
>> a device from the outside.  I followed the wiki that suggest  
>> creating a separate profile to handle external device connections.
>>
>> Here?s my external xml file
>> <profile name="gexternal5090">
>>   <!-- This profile is only for outbound registrations to providers  
>> -->
>>   <gateways>
>>     <X-PRE-PROCESS cmd="include" data="external/*.xml"/>
>>   </gateways>
>>   <aliases>
>>     <alias name="outbound"/>
>>   </aliases>
>>   <domains>
>>     <domain name="$${domain}" parse="true"/>
>>   </domains>
>>   <settings>
>>     <param name="debug" value="0"/>
>>     <param name="sip-trace" value="no"/>
>>     <param name="rfc2833-pt" value="101"/>
>>     <param name="sip-port" value="5090"/>
>>     <param name="dialplan" value="XML"/>
>>     <param name="context" value="public"/>
>>     <param name="dtmf-duration" value="100"/>
>>     <param name="codec-prefs" value="$${outbound_codec_prefs}"/>
>>     <param name="hold-music" value="$${moh_uri}"/>
>>     <param name="use-rtp-timer" value="true"/>
>>     <param name="rtp-timer-name" value="soft"/>
>>     <param name="manage-presence" value="false"/>
>>     <param name="inbound-codec-negotiation" value="generous"/>
>>     <param name="nonce-ttl" value="60"/>
>>     <param name="auth-calls" value="false"/>
>>     <param name="rtp-timeout-sec" value="1800"/>
>>     <param name="rtp-ip" value="$${local_ip_v4}"/>
>>     <param name="sip-ip" value="$${local_ip_v4}"/>
>>     <param name="ext-rtp-ip" value="71.133.39.219"/>
>>     <param name="ext-sip-ip" value="71.133.39.219"/>
>>     <param name="rtp-timeout-sec" value="300"/>
>>     <param name="rtp-hold-timeout-sec" value="1800"/>
>>   </settings>
>> </profile>
>>
>> I did a siptrace on the incoming connection attempt
>>    
>> --- 
>> ---------------------------------------------------------------------
>>    REGISTER sip:71.133.39.219:5090 SIP/2.0
>>    Via: SIP/2.0/UDP x.x.x.x:5090;branch=z9hG4bKvvaun9r6.1u8ej
>>    Max-Forwards: 70
>>    From: <sip:1001 at 71.133.39.219>;tag=i9w3pt4bm4tpsl
>>    To: <sip:1001 at 71.133.39.219>
>>    Call-ID: fo33ia7sk6qy9vnk6h
>>    CSeq: 75203 REGISTER
>>    Contact: <sip:1001 at x.x.x.x:5090;line=20880>
>>    Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER,  
>> SUBSCRIBE, NOTIFY, MESSAGE, INFO, PRACK
>>    Expires: 600
>>    User-Agent: snom-m3-SIP/02.02 (MAC=0004132AD0BE; HW=255)
>>    Content-Length: 0
>>
>>     
>> --- 
>> ---------------------------------------------------------------------
>> send 591 bytes to udp/[x.x.x.x]:5090 at 01:26:47.109165:
>>     
>> --- 
>> ---------------------------------------------------------------------
>>    SIP/2.0 401 Unauthorized
>>    Via: SIP/2.0/UDP  
>> 10.0.10.231:5090;branch=z9hG4bKvvaun9r6.1u8ej;received=207.47.31.108
>>    From: <sip:1001 at 71.133.39.219>;tag=i9w3pt4bm4tpsl
>>    To: <sip:1001 at 71.133.39.219>;tag=Zc7mBaBN723Bp
>>    Call-ID: fo33ia7sk6qy9vnk6h
>>    CSeq: 75203 REGISTER
>>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,  
>> REGISTER, REFER, NOTIFY
>>    Supported: timer, precondition, path, replaces
>>    WWW-Authenticate: Digest realm="71.133.39.219",  
>> nonce="bd1ecf94-5589-11df-b3f4-c1a375adf51a", algorithm=MD5,  
>> qop="auth"
>>    Content-Length: 0
>>
>>     
>> --- 
>> ---------------------------------------------------------------------
>>
>> Thoughts?
>> Thanks
>> Sean
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From lloyd.aloysius at gmail.com  Sun May  2 11:12:18 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sun, 2 May 2010 14:12:18 -0400
Subject: [Freeswitch-users] Emacs/Vim - XML Editing Settings
Message-ID: <t2j8a19bf2e1005021112v54666e8fh281cd506ef21e47d@mail.gmail.com>

Hi All,

The default xml configuration files indent very nicely. What is the setting
used to edit the default xml  configuration files.

Could you please share the settings either for emacs or vim. I prefer to use
emacs.

Thanks in advance.
Lloyd
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From brian at freeswitch.org  Sun May  2 11:37:21 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 2 May 2010 13:37:21 -0500
Subject: [Freeswitch-users] nat problem
In-Reply-To: <q2g1fdf92721005020122n23fc1f6x89bfa9532123f520@mail.gmail.com>
References: <q2g1fdf92721005020122n23fc1f6x89bfa9532123f520@mail.gmail.com>
Message-ID: <1BA68902-B2FE-4F2C-8D8C-FE90933C13C6@freeswitch.org>

You haven't set the local-network-acl on the profile and if you have ext-rtp-ip and ext-sip-ip set and you're traversing nat also you'll need to prefix them with autonat:

/b

On May 2, 2010, at 3:22 AM, babak yakhchali wrote:

> Hi
> I got two 7941 ip phones connected to freeswitch (ip phone)192.168.22.57 -> 192.168.22.1 -> 192.168.11.30(fs)
> they can register fine but I can not call from ip phone to ip phone. when I use 3cx softphone in the same conditions I can call from ip phone to 3cx but not from 3cx to ip phone this is result of  "sofia status profile internal" and when I get pcap on fs when invite is sent to ip phones nothing comes back 
> 



From brian at freeswitch.org  Sun May  2 11:38:14 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 2 May 2010 13:38:14 -0500
Subject: [Freeswitch-users] Emacs/Vim - XML Editing Settings
In-Reply-To: <t2j8a19bf2e1005021112v54666e8fh281cd506ef21e47d@mail.gmail.com>
References: <t2j8a19bf2e1005021112v54666e8fh281cd506ef21e47d@mail.gmail.com>
Message-ID: <084B5103-E47C-4BBF-810C-8A3CB365C89F@freeswitch.org>

nxml mode in emacs.

/b
see: support-d/install-cc-mode.sh

On May 2, 2010, at 1:12 PM, Aloysius Lloyd wrote:

> Could you please share the settings either for emacs or vim. I prefer to use emacs.



From lloyd.aloysius at gmail.com  Sun May  2 12:00:04 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Sun, 2 May 2010 15:00:04 -0400
Subject: [Freeswitch-users] Emacs/Vim - XML Editing Settings
In-Reply-To: <084B5103-E47C-4BBF-810C-8A3CB365C89F@freeswitch.org>
References: <t2j8a19bf2e1005021112v54666e8fh281cd506ef21e47d@mail.gmail.com> 
	<084B5103-E47C-4BBF-810C-8A3CB365C89F@freeswitch.org>
Message-ID: <k2u8a19bf2e1005021200naba82e5bk98079d840921c3e3@mail.gmail.com>

Thanks Brian. The nxml mode is working great.

Lloyd


On Sun, May 2, 2010 at 2:38 PM, Brian West <brian at freeswitch.org> wrote:

> nxml mode in emacs.
>
> /b
> see: support-d/install-cc-mode.sh
>
> On May 2, 2010, at 1:12 PM, Aloysius Lloyd wrote:
>
> > Could you please share the settings either for emacs or vim. I prefer to
> use emacs.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From babak.freeswitch at gmail.com  Sun May  2 12:37:25 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Mon, 3 May 2010 00:07:25 +0430
Subject: [Freeswitch-users] nat problem
In-Reply-To: <1BA68902-B2FE-4F2C-8D8C-FE90933C13C6@freeswitch.org>
References: <q2g1fdf92721005020122n23fc1f6x89bfa9532123f520@mail.gmail.com>
	<1BA68902-B2FE-4F2C-8D8C-FE90933C13C6@freeswitch.org>
Message-ID: <v2k1fdf92721005021237nb7f77d01k51764c74c72a4748@mail.gmail.com>

I'm using the internal profile of 1.0.6 without any changes. and in the file
I got:
<param name="local-network-acl" value="localnet.auto"/>
and
<param name="ext-rtp-ip" value="auto-nat"/>
    <param name="ext-sip-ip" value="auto-nat"/>
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From brian at freeswitch.org  Sun May  2 12:45:57 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 2 May 2010 14:45:57 -0500
Subject: [Freeswitch-users] nat problem
In-Reply-To: <v2k1fdf92721005021237nb7f77d01k51764c74c72a4748@mail.gmail.com>
References: <q2g1fdf92721005020122n23fc1f6x89bfa9532123f520@mail.gmail.com>
	<1BA68902-B2FE-4F2C-8D8C-FE90933C13C6@freeswitch.org>
	<v2k1fdf92721005021237nb7f77d01k51764c74c72a4748@mail.gmail.com>
Message-ID: <78C5F359-E703-4D62-954A-D8802127950A@freeswitch.org>

You will have to change local-network-acl to include more than just 192.168.11.0/24

Your phones are on 192.168.22.0/24 so the ACL isn't matching please change it to rfc1918.auto

/b

On May 2, 2010, at 2:37 PM, babak yakhchali wrote:

> I'm using the internal profile of 1.0.6 without any changes. and in the file
> I got:
> <param name="local-network-acl" value="localnet.auto"/>
> and
> <param name="ext-rtp-ip" value="auto-nat"/>
>     <param name="ext-sip-ip" value="auto-nat"/>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From babak.freeswitch at gmail.com  Sun May  2 12:59:35 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Mon, 3 May 2010 00:29:35 +0430
Subject: [Freeswitch-users] nat problem
In-Reply-To: <78C5F359-E703-4D62-954A-D8802127950A@freeswitch.org>
References: <q2g1fdf92721005020122n23fc1f6x89bfa9532123f520@mail.gmail.com>
	<1BA68902-B2FE-4F2C-8D8C-FE90933C13C6@freeswitch.org>
	<v2k1fdf92721005021237nb7f77d01k51764c74c72a4748@mail.gmail.com>
	<78C5F359-E703-4D62-954A-D8802127950A@freeswitch.org>
Message-ID: <g2n1fdf92721005021259oe629cfd4ne901f3236188d5bb@mail.gmail.com>

thanks but what about 3cx soft phones? they are on 192.168.22.0/24 too
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From david.ponzone at gmail.com  Sun May  2 13:41:18 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Sun, 2 May 2010 22:41:18 +0200
Subject: [Freeswitch-users] nat problem
In-Reply-To: <g2n1fdf92721005021259oe629cfd4ne901f3236188d5bb@mail.gmail.com>
References: <q2g1fdf92721005020122n23fc1f6x89bfa9532123f520@mail.gmail.com>
	<1BA68902-B2FE-4F2C-8D8C-FE90933C13C6@freeswitch.org>
	<v2k1fdf92721005021237nb7f77d01k51764c74c72a4748@mail.gmail.com>
	<78C5F359-E703-4D62-954A-D8802127950A@freeswitch.org>
	<g2n1fdf92721005021259oe629cfd4ne901f3236188d5bb@mail.gmail.com>
Message-ID: <B0C6BC7E-0233-40AA-B0B8-7903A8907DAA@gmail.com>

Babak,

I think the issue is that the network topology between the phones and  
FS is not what you think.

 From FS, if you can ping a phone's IP, it means they are not behind  
NAT.
So you should do what Brian asked, so that FS doesn't think they are  
behind NAT.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 02/05/2010 ? 21:59, babak yakhchali a ?crit :

> thanks but what about 3cx soft phones? they are on 192.168.22.0/24  
> too _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From vfclists at googlemail.com  Sun May  2 14:01:31 2010
From: vfclists at googlemail.com (Frank Church)
Date: Sun, 2 May 2010 22:01:31 +0100
Subject: [Freeswitch-users] mod_xml_cdr is not logging at all.
In-Reply-To: <EB5AE6C1-8718-4190-8745-319631D65D99@gmail.com>
References: <n2z7abab2411005010051ye18ea1f2s5e5cf2e57158d09c@mail.gmail.com>
	<h2v7abab2411005011400u3dcab078zba3e50d5fc6b3418@mail.gmail.com>
	<4BDC9D2C.9090507@todandlorna.com>
	<o2t7abab2411005011511ha22b18dfs1b599c3c74de8e87@mail.gmail.com>
	<EB5AE6C1-8718-4190-8745-319631D65D99@gmail.com>
Message-ID: <k2x7abab2411005021401x12955ef7s5c342d39cda54b3e@mail.gmail.com>

I have tested it now and the <param name="log-http-and-disk" value="true"/>
is necessary for logging to disk. Logging to disk doesn't work without it
even if you set logging to the default values.

On 1 May 2010 23:26, David Ponzone <david.ponzone at gmail.com> wrote:

> Frank,
>
> In my config, I have:
>
> <param name="log-dir" value=""/>
>
> This should be enough to get the XML CDRs written to
> $freeswitch_dir/log/xml_cdr/
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 02/05/2010 ? 00:11, Frank Church a ?crit :
>
> mox_xml_cdr is working fine. I am sending the XML to webserver and saving
> the information into a database. It is the file logging which is not working
> at all.
>
> What I don't know is if the file path I am using is correct.
>
> cdr_csv works fine and it is a Windows system I don't think it is a file
> permisions problem.
>
> On 1 May 2010 22:29, Tod Hansmann <freeswitch.org at todandlorna.com> wrote:
>
>>  Directory permissions maybe?  *shrug*  Could use some more information on
>> what steps you've taken. =c)
>>
>> -Tod Hansmann
>>
>>
>> On 5/1/2010 3:00 PM, Frank Church wrote:
>>
>> Any relief with this question?
>> It is beginning to feel rather lonely.
>>
>> On 1 May 2010 08:51, Frank Church <vfclists at googlemail.com> wrote:
>>
>>> Mod_xml_cdr is not logging anything at all. I have created an additional
>>> directory called logs, and it is still not working.
>>> cdr_csv is logging fine.
>>>
>>> Below is the current file, are there any faults in it?
>>> Which directory does {prefix} point to?
>>>
>>> My current verson is 17408m on Windows
>>>
>>> <configuration name="xml_cdr.conf" description="XML CDR CURL logger">
>>>>   <settings>
>>>>     <param name="url" value="http://192.168.1.20:8101/"/>
>>>>     <param name="log-dir" value="log/xml_cdr"/>
>>>>     <param name="log-b-leg" value="false"/>
>>>>     <param name="prefix-a-leg" value="true"/>
>>>>     <param name="encode" value="true"/>
>>>>      <param name="err-log-dir" value="log/xml_cdr"/> -->
>>>>   </settings>
>>>> </configuration>
>>>>
>>>
>>>
>>> --
>>> Frank Church
>>>
>>> =======================
>>> http://devblog.brahmancreations.com
>>>
>>
>>
>>
>> --
>> Frank Church
>>
>> =======================
>> http://devblog.brahmancreations.com
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Frank Church
>
> =======================
> http://devblog.brahmancreations.com
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Frank Church

=======================
http://devblog.brahmancreations.com
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From mike at jerris.com  Sun May  2 21:04:48 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 3 May 2010 00:04:48 -0400
Subject: [Freeswitch-users] run error after building in vs 2008!
In-Reply-To: <x2z367751821005011229g3b6c217kbda766daac32a44d@mail.gmail.com>
References: <i2v1fdf92721004302253l7dbdf254p1ff2b3884ec5a4d4@mail.gmail.com>
	<l2o367751821005010612j46281f6fg17629c38bcd07f4@mail.gmail.com>
	<n2i1fdf92721005011131o221b6dd4sf483e00613963d5@mail.gmail.com>
	<x2z367751821005011229g3b6c217kbda766daac32a44d@mail.gmail.com>
Message-ID: <E141A983-9EC5-4C62-8D17-675DC0A67009@jerris.com>

This was just added in :

commit f1430d521a767a13035d6d8e96686564552194fd
Author: Anthony Minessale <anthm at freeswitch.org>
Date:   Fri Apr 30 15:01:32 2010 -0500

    fix switch_get_addr to work with v6 properly

to fix ipv6 support.  We welcome a patch to restore xp support.

Mike

On May 1, 2010, at 3:29 PM, Phillip Jones wrote:

> Oh yeah - looks like that call is not supported in older os like XP.
> 
> http://social.msdn.microsoft.com/Forums/en-US/vcgeneral/thread/e40465f2-41b7-4243-ad33-15ae9366f4e6
> 
> FreeSWITCH does support XP though - http://wiki.freeswitch.org/wiki/Users_Guide_Introduction
> 
> So I don't know.
> 
> On Sat, May 1, 2010 at 2:31 PM, babak yakhchali <babak.freeswitch at gmail.com> wrote:
> ya I've got that in place
> but as I searched the web they say it's related to windows version (xp vista . . .)

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From brian at freeswitch.org  Sun May  2 21:14:14 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 2 May 2010 23:14:14 -0500
Subject: [Freeswitch-users] Aastra and SCA
In-Reply-To: <021d01cae8d2$fc999420$f5ccbc60$@maly@molcs.org>
References: <021d01cae8d2$fc999420$f5ccbc60$@maly@molcs.org>
Message-ID: <2A6C7EC2-913D-417E-ADD6-E3A72C6AA6E6@freeswitch.org>

Give up their exists a bug that prevents it from working on any aastra.  Polycom and Cisco SPA work flawless.

/b

Sent from my iPad

On Apr 30, 2010, at 9:07 PM, "Mark Maly" <mark.maly at molcs.org> wrote:

> Hi,
> 
>  
> 
> I?ve tried to patiently figure this out by reading the wiki and this list.  Unfortunately, I?ve been unable to get it right. 
> 
>  
> 
> I have 2 Aastra 6731is and a 51i and trying to get SCA working.  I?m experiencing problems similar to the Cisco thread from last month ? outgoing calls implement SCA well.  Incoming calls ring all lines and appearances work, but when one phone is answered, the line appearances are removed from the remaining phones.  I am not attempting to use any DNS.
> 
>  
> 
> My configuration has all three phones plus FS on a local LAN.  Nothing too fancy.  Each line is configured for Broadsoft SCA and SCA bridging  is enabled globally for the phones.
> 
>  
> 
> Trying to update/replace an old phone at my church.
> 
>  
> 
> Any help would be greatly appreciated.
> 
>  
> 
> Thanks, Mark
> 
> Mark.maly at molcs.org
> 
>  
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From mike at jerris.com  Sun May  2 21:18:07 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 3 May 2010 00:18:07 -0400
Subject: [Freeswitch-users] Aastra and SCA
In-Reply-To: <021d01cae8d2$fc999420$f5ccbc60$@maly@molcs.org>
References: <021d01cae8d2$fc999420$f5ccbc60$@maly@molcs.org>
Message-ID: <21E6CEC8-59C9-4E2A-92ED-D7B93286C5DC@jerris.com>

Aastra has a broken SCA implementation.  We have been working with them to resolve it but as of yet they have failed to deliver a working firmware for their phones.

Mike

On Apr 30, 2010, at 10:07 PM, Mark Maly wrote:

> Hi,
>  
> I?ve tried to patiently figure this out by reading the wiki and this list.  Unfortunately, I?ve been unable to get it right. 
>  
> I have 2 Aastra 6731is and a 51i and trying to get SCA working.  I?m experiencing problems similar to the Cisco thread from last month ? outgoing calls implement SCA well.  Incoming calls ring all lines and appearances work, but when one phone is answered, the line appearances are removed from the remaining phones.  I am not attempting to use any DNS.
>  
> My configuration has all three phones plus FS on a local LAN.  Nothing too fancy.  Each line is configured for Broadsoft SCA and SCA bridging  is enabled globally for the phones.
>  
> Trying to update/replace an old phone at my church.

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From mike at jerris.com  Sun May  2 21:19:18 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 3 May 2010 00:19:18 -0400
Subject: [Freeswitch-users] Accessing SQLITE core.db
In-Reply-To: <p2qc96b95651004290347la4df1c24t8c02d2d9dac9de74@mail.gmail.com>
References: <q2kc96b95651004290310va9fe1cdfn133c4073dd82ec0@mail.gmail.com>
	<x2g27b0d5bc1004290327t8e8d59ffn59dd994032e60b7@mail.gmail.com>
	<p2qc96b95651004290347la4df1c24t8c02d2d9dac9de74@mail.gmail.com>
Message-ID: <548A7285-10B3-40FE-909A-C0D0F2DAADAC@jerris.com>

I highly recommend using an external db via odbc if you will access it like this.

On Apr 29, 2010, at 6:47 AM, Saeed Ahmed wrote:

> Since i am using it on Centos, so it seems that its safe to send 'select' query to core.db.
> 
> On Thu, Apr 29, 2010 at 12:27 PM, Justin B Newman <justin at ejtown.org> wrote:
> On Thu, Apr 29, 2010 at 6:10 AM, Saeed Ahmed <saeedahmad1981 at gmail.com> wrote:
> >
> > Is it safe to access sqlite db using PHP, when there are live calls on FS?
> >
> > I am just sending selects to 'channels' table to view live calls.
> >
> 
> http://www.sqlite.org/faq.html#q5
> 

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From mike at jerris.com  Sun May  2 22:04:15 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 3 May 2010 01:04:15 -0400
Subject: [Freeswitch-users] Build fails on Suse
In-Reply-To: <4BD96479020000F000017787@firewall.thesummit-grp.com>
References: <4BD96479020000F000017787@firewall.thesummit-grp.com>
Message-ID: <CF93C845-8249-4B80-9990-65DC174E1B10@jerris.com>

I think all of these issues have been fixed in git head already.

Mike

On Apr 29, 2010, at 10:50 AM, Matt White wrote:

> I'm using the opensuse build service to create rpm's for Suse.
> 
> The build fails with the following error below.  I think its due to the gcc 4.3 used on SLE11 as I can't replicate it in older versions
> 
> 
> I: Expression compares a char* pointer with a string literal.
>    Usually a strcmp() was intended by the programmer
> E: freeswitch stringcompare strings/apr_snprintf.c:1261
> 
> Any thoughts?  I'm using the nightly snapshot.  

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From mike at jerris.com  Sun May  2 22:06:31 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 3 May 2010 01:06:31 -0400
Subject: [Freeswitch-users] mod_xml_cdr error log is not working
In-Reply-To: <g2p7abab2411004291856s2c2d5018g103f33cf2be9dd4b@mail.gmail.com>
References: <g2p7abab2411004291856s2c2d5018g103f33cf2be9dd4b@mail.gmail.com>
Message-ID: <DD0949EC-0C93-4A8F-9EA4-C6A669AE93FF@jerris.com>

the defaults in windows are all relative to the running freeswitch base dir, unless you explicitly set them.  there may be some issue with trailing vs no trailing path seperator when you configure them.

Mike

On Apr 29, 2010, at 9:56 PM, Frank Church wrote:

> I have set the err-log-dir and log-dr in xml_cdr.conf.xml but the logs
> are not working. The system is a windows system and I wonder if the
> defaults for windows are different.
> 
> I have logs/xml_cdr  in addition to the  log/xml_cdr in the
> c:\freeswitch directory but Freeswitch can't find them.
> 
> Logs snippet
> =========
> 2010-04-30 02:41:02.984375 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/1811 at 192.168.1.133) Running State Change CS_REPORTING
> 2010-04-30 02:41:02.984375 [DEBUG] switch_core_state_machine.c:590
> (sofia/internal/1811 at 192.168.1.133) State REPORTING
> 2010-04-30 02:41:03.812500 [ERR] mod_xml_cdr.c:359 Got error [0]
> posting to web server [http://192.168.1.20:8132/]
> 2010-04-30 02:41:03.812500 [ERR] mod_xml_cdr.c:366 Retry will be with
> url [http://192.168.1.20:8132/]
> 2010-04-30 02:41:03.812500 [ERR] mod_xml_cdr.c:377 Unable to post to
> web server, writing to file
> 2010-04-30 02:41:03.812500 [ERR] mod_xml_cdr.c:399 Error![No such file
> or directory]
> 



From babak.freeswitch at gmail.com  Sun May  2 22:06:54 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Mon, 3 May 2010 09:36:54 +0430
Subject: [Freeswitch-users] nat problem
In-Reply-To: <B0C6BC7E-0233-40AA-B0B8-7903A8907DAA@gmail.com>
References: <q2g1fdf92721005020122n23fc1f6x89bfa9532123f520@mail.gmail.com>
	<1BA68902-B2FE-4F2C-8D8C-FE90933C13C6@freeswitch.org>
	<v2k1fdf92721005021237nb7f77d01k51764c74c72a4748@mail.gmail.com>
	<78C5F359-E703-4D62-954A-D8802127950A@freeswitch.org>
	<g2n1fdf92721005021259oe629cfd4ne901f3236188d5bb@mail.gmail.com>
	<B0C6BC7E-0233-40AA-B0B8-7903A8907DAA@gmail.com>
Message-ID: <n2g1fdf92721005022206l18bff8c0pa4dda0e541ede54f@mail.gmail.com>

Brian and David
Thank u so much for answering my question. my problem is solved. thank u a
lot u were right, the problem was with local acl and now it's solved
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From vfclists at googlemail.com  Mon May  3 03:43:47 2010
From: vfclists at googlemail.com (Frank Church)
Date: Mon, 3 May 2010 11:43:47 +0100
Subject: [Freeswitch-users] mod_xml_cdr error log is not working
In-Reply-To: <DD0949EC-0C93-4A8F-9EA4-C6A669AE93FF@jerris.com>
References: <g2p7abab2411004291856s2c2d5018g103f33cf2be9dd4b@mail.gmail.com>
	<DD0949EC-0C93-4A8F-9EA4-C6A669AE93FF@jerris.com>
Message-ID: <l2z7abab2411005030343xdbe1827coc7e531214aa78864@mail.gmail.com>

I have received some help about it in
http://lists.freeswitch.org/pipermail/freeswitch-users/2010-May/057453.html.

It requires an additional undocumented parameter in mod_xml_cdr.conf, though
that may not have been necessary in earlier versions.


On 3 May 2010 06:06, Michael Jerris <mike at jerris.com> wrote:

> the defaults in windows are all relative to the running freeswitch base
> dir, unless you explicitly set them.  there may be some issue with trailing
> vs no trailing path seperator when you configure them.
>
> Mike
>
> On Apr 29, 2010, at 9:56 PM, Frank Church wrote:
>
> > I have set the err-log-dir and log-dr in xml_cdr.conf.xml but the logs
> > are not working. The system is a windows system and I wonder if the
> > defaults for windows are different.
> >
> > I have logs/xml_cdr  in addition to the  log/xml_cdr in the
> > c:\freeswitch directory but Freeswitch can't find them.
> >
> > Logs snippet
> > =========
> > 2010-04-30 02:41:02.984375 [DEBUG] switch_core_state_machine.c:314
> > (sofia/internal/1811 at 192.168.1.133) Running State Change CS_REPORTING
> > 2010-04-30 02:41:02.984375 [DEBUG] switch_core_state_machine.c:590
> > (sofia/internal/1811 at 192.168.1.133) State REPORTING
> > 2010-04-30 02:41:03.812500 [ERR] mod_xml_cdr.c:359 Got error [0]
> > posting to web server [http://192.168.1.20:8132/]
> > 2010-04-30 02:41:03.812500 [ERR] mod_xml_cdr.c:366 Retry will be with
> > url [http://192.168.1.20:8132/]
> > 2010-04-30 02:41:03.812500 [ERR] mod_xml_cdr.c:377 Unable to post to
> > web server, writing to file
> > 2010-04-30 02:41:03.812500 [ERR] mod_xml_cdr.c:399 Error![No such file
> > or directory]
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Frank Church

=======================
http://devblog.brahmancreations.com
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From kawarod at laposte.net  Mon May  3 06:39:59 2010
From: kawarod at laposte.net (Rod.)
Date: Mon, 03 May 2010 17:39:59 +0400
Subject: [Freeswitch-users] Registration ODBC feeded by another registrar
 proxy
Message-ID: <4BDED22F.20205@laposte.net>

Hi list,

was playing with FS 1.0.6 and trying to test the registration 
performance of FS. (Yes I know FS is more suited as a B2BUA, but please 
read further :p)

So I did the following:
     - generate one xml file with 20 000 user account like this:
<include>
<user id="1">
<params>
<param name="password" value="1234"/>
</params>
</user>
<user id="2">
<params>
<param name="password" value="1234"/>
</params>
             ...

Then I used Sipp to test how many registration per second could be fired 
to the server (quad core 2.83Ghz).
I setup ulimit variables, and disable nat.

I got this:
     - using SQL Lite: unable to get higher than 80 registrations per 
second (in fact it's less than this number but didn't test too much this 
setup), I see a lot of retransmission in Sipp
     - using SQL Lite in ramdisk (tmpfs): OK with 80 registrations per 
second but not much
     - using ODBC and mysql: 130 reg/sec is OK

With ODBC, above 150 reg/sec I see that FS is stalled to 100-110% CPU, I 
think it's because I'm using only one SIP profile and that SOFIA is 
monothreaded for this SIP profile.
If I'd like to register every 60sec, the server has to support at least 
more than 300 registration per second.

So I'm wondering if I could setup something like this:
     - use another SIP Proxy as a registrar and feed the ODBC 
"sip_registration database" of FS
     - FS will be able to use this database to setup a call
     - use FS as the outbound proxy for call routing

But what about the user params that have been setup in the xml file 
above. I think that FS loads the user params each time a user is registered.

Comments and advices are welcome.

regards,
rod.




From janvb at live.com  Mon May  3 08:14:07 2010
From: janvb at live.com (Jan Berger)
Date: Mon, 3 May 2010 17:14:07 +0200
Subject: [Freeswitch-users] Registration ODBC feeded by another
 registrar proxy
In-Reply-To: <4BDED22F.20205@laposte.net>
References: <4BDED22F.20205@laposte.net>
Message-ID: <SNT135-w38911E36E11BBC4B429F77D2F20@phx.gbl>


hi Rod,

 

What is the objective and intended usage of this ?

 

Registering loads of accounts every sec sounds like a very bad design to me, so it might be wise to visit your design and discuss what you want to achieve.

 

 

Jan
 
> Date: Mon, 3 May 2010 17:39:59 +0400
> From: kawarod at laposte.net
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Registration ODBC feeded by another registrar proxy
> 
> Hi list,
> 
> was playing with FS 1.0.6 and trying to test the registration 
> performance of FS. (Yes I know FS is more suited as a B2BUA, but please 
> read further :p)
> 
> So I did the following:
> - generate one xml file with 20 000 user account like this:
> <include>
> <user id="1">
> <params>
> <param name="password" value="1234"/>
> </params>
> </user>
> <user id="2">
> <params>
> <param name="password" value="1234"/>
> </params>
> ...
> 
> Then I used Sipp to test how many registration per second could be fired 
> to the server (quad core 2.83Ghz).
> I setup ulimit variables, and disable nat.
> 
> I got this:
> - using SQL Lite: unable to get higher than 80 registrations per 
> second (in fact it's less than this number but didn't test too much this 
> setup), I see a lot of retransmission in Sipp
> - using SQL Lite in ramdisk (tmpfs): OK with 80 registrations per 
> second but not much
> - using ODBC and mysql: 130 reg/sec is OK
> 
> With ODBC, above 150 reg/sec I see that FS is stalled to 100-110% CPU, I 
> think it's because I'm using only one SIP profile and that SOFIA is 
> monothreaded for this SIP profile.
> If I'd like to register every 60sec, the server has to support at least 
> more than 300 registration per second.
> 
> So I'm wondering if I could setup something like this:
> - use another SIP Proxy as a registrar and feed the ODBC 
> "sip_registration database" of FS
> - FS will be able to use this database to setup a call
> - use FS as the outbound proxy for call routing
> 
> But what about the user params that have been setup in the xml file 
> above. I think that FS loads the user params each time a user is registered.
> 
> Comments and advices are welcome.
> 
> regards,
> rod.
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  
_________________________________________________________________
Your E-mail and More On-the-Go. Get Windows Live Hotmail Free.
https://signup.live.com/signup.aspx?id=60969
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From pjintheusa at gmail.com  Mon May  3 08:21:23 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Mon, 3 May 2010 11:21:23 -0400
Subject: [Freeswitch-users] Registration ODBC feeded by another
	registrar proxy
In-Reply-To: <4BDED22F.20205@laposte.net>
References: <4BDED22F.20205@laposte.net>
Message-ID: <s2r367751821005030821q488ecc4fz14b988a8d3d20582@mail.gmail.com>

Rod - interesting. I am also thinking about the best architecture as far as
registrations goes.

So things that spring to my mind include:

- using more that one FS box to spread the load - basically create a FS
cluster - with OpenSIPS in front to load balance
- use *mod_xml_curl *to pull /directory info from one source
- possibly using OpenSIPS to handle registrations (although I am not sure
how to get around the single point of failure issue there)



On Mon, May 3, 2010 at 9:39 AM, Rod. <kawarod at laposte.net> wrote:

> Hi list,
>
> was playing with FS 1.0.6 and trying to test the registration
> performance of FS. (Yes I know FS is more suited as a B2BUA, but please
> read further :p)
>
> So I did the following:
>     - generate one xml file with 20 000 user account like this:
> <include>
> <user id="1">
> <params>
> <param name="password" value="1234"/>
> </params>
> </user>
> <user id="2">
> <params>
> <param name="password" value="1234"/>
> </params>
>             ...
>
> Then I used Sipp to test how many registration per second could be fired
> to the server (quad core 2.83Ghz).
> I setup ulimit variables, and disable nat.
>
> I got this:
>     - using SQL Lite: unable to get higher than 80 registrations per
> second (in fact it's less than this number but didn't test too much this
> setup), I see a lot of retransmission in Sipp
>     - using SQL Lite in ramdisk (tmpfs): OK with 80 registrations per
> second but not much
>     - using ODBC and mysql: 130 reg/sec is OK
>
> With ODBC, above 150 reg/sec I see that FS is stalled to 100-110% CPU, I
> think it's because I'm using only one SIP profile and that SOFIA is
> monothreaded for this SIP profile.
> If I'd like to register every 60sec, the server has to support at least
> more than 300 registration per second.
>
> So I'm wondering if I could setup something like this:
>     - use another SIP Proxy as a registrar and feed the ODBC
> "sip_registration database" of FS
>     - FS will be able to use this database to setup a call
>     - use FS as the outbound proxy for call routing
>
> But what about the user params that have been setup in the xml file
> above. I think that FS loads the user params each time a user is
> registered.
>
> Comments and advices are welcome.
>
> regards,
> rod.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From hungngm1987 at gmail.com  Mon May  3 00:20:24 2010
From: hungngm1987 at gmail.com (hung nguyen)
Date: Mon, 3 May 2010 14:20:24 +0700
Subject: [Freeswitch-users] Need help on realtime integration Freeswitch
	with Opensips.
Message-ID: <s2jd9f1e4661005030020qb61f7128o57fe42aaa2cfb3c3@mail.gmail.com>

Hi list.
I have deployed cluster FS with share DB - using ODBC in core.
I think it is great if we can deploy  realtime integration Freeswitch with
Opensips. With this, Opensips will act as register,proxy server and load
balancer in front of FS boxs.
Anybody have same ideal and done this.
Tks.
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From josephcrivello at gmail.com  Mon May  3 04:36:23 2010
From: josephcrivello at gmail.com (Joseph Crivello)
Date: Mon, 3 May 2010 06:36:23 -0500
Subject: [Freeswitch-users] Trouble With 'transfer' DP After a 'bridge' on
	Async Event Socket
Message-ID: <07a101caeab4$dd343700$979ca500$@com>

Hello All,

I am currently working on a conference bridge based on FreeSWITCH that
allows some conference members to dial out to a prompted number.

Currently my implementation binds * in conference.xml to the "transfer"
action, leading to a dial plan extension that runs an async full outgoing
event socket. My application serving the event socket does the following:

1) Verifies the user is allowed to dial out, and writes a record to the
database.
2) Retrieves the number to be dialed with 'play_and_get_digits'
3) Bridges the call to the dialed number
4) Waits for the a-leg to press star
5) Uses 'transfer' to transfer both legs back to the conference 

Everything works up to step #5, which does nothing. 

There is no NOTICE outputted to the console indicating that a transfer is
taking place and neither leg is rejoined to the conference.

I get a normal event back after the transfer sendmsg:

Event-Name: SOCKET_DATA
Content-Type: command/reply
Reply-Text: %2BOK

If I subscribe to all events before running the transfer and receive events
in a loop, I receive nothing but routine RE_SCHEDULE and CALL_UPDATE events
after the transfer.

I am using ESL to communicate with FreeSWITCH, and prior to running any of
steps #1-5 I set the event lock to true. The line from the dial plan
extension that calls my application follows:

<action application="socket" data="172.18.11.1:8001 async full" />

Any ideas? Perhaps I'm "doing it wrong"? I am very new to FS.

Thanks!
Joe Crivello




From josephcrivello at gmail.com  Mon May  3 07:27:55 2010
From: josephcrivello at gmail.com (Joe Crivello)
Date: Mon, 3 May 2010 09:27:55 -0500
Subject: [Freeswitch-users] Fwd: Trouble With 'transfer' DP After a 'bridge'
	on Async Event Socket
In-Reply-To: <07a101caeab4$dd343700$979ca500$@com>
References: <07a101caeab4$dd343700$979ca500$@com>
Message-ID: <x2rf1b753861005030727h2aa52e71r1e23257c28213aed@mail.gmail.com>

Strange.. my previous message got lost somehow..

---------- Forwarded message ----------
From: Joseph Crivello <josephcrivello at gmail.com>
Date: Mon, May 3, 2010 at 6:36 AM
Subject: Trouble With 'transfer' DP After a 'bridge' on Async Event Socket
To: freeswitch-users at lists.freeswitch.org

Hello All,

I am currently working on a conference bridge based on FreeSWITCH that
allows some conference members to dial out to a prompted number.

Currently my implementation binds * in conference.xml to the "transfer"
action, leading to a dial plan extension that runs an async full outgoing
event socket. My application serving the event socket does the following:

1) Verifies the user is allowed to dial out, and writes a record to the
database.
2) Retrieves the number to be dialed with 'play_and_get_digits'
3) Bridges the call to the dialed number
4) Waits for the a-leg to press star
5) Uses 'transfer' to transfer both legs back to the conference

Everything works up to step #5, which does nothing.

There is no NOTICE outputted to the console indicating that a transfer is
taking place and neither leg is rejoined to the conference.

I get a normal event back after the transfer sendmsg:

Event-Name: SOCKET_DATA
Content-Type: command/reply
Reply-Text: %2BOK

If I subscribe to all events before running the transfer and receive events
in a loop, I receive nothing but routine RE_SCHEDULE and CALL_UPDATE events
after the transfer.

I am using ESL to communicate with FreeSWITCH, and prior to running any of
steps #1-5 I set the event lock to true. The line from the dial plan
extension that calls my application follows:

<action application="socket" data="172.18.11.1:8001 async full" />

Any ideas? Perhaps I'm "doing it wrong"? I am very new to FS.

Thanks!
Joe Crivello
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From josephcrivello at gmail.com  Mon May  3 07:33:01 2010
From: josephcrivello at gmail.com (Joe Crivello)
Date: Mon, 3 May 2010 09:33:01 -0500
Subject: [Freeswitch-users] Trouble With Transfer and Async Event Socket
Message-ID: <n2sf1b753861005030733u73a51a2fh4fae23812f5089@mail.gmail.com>

Hello All,

I am currently working on a conference bridge based on FreeSWITCH that
allows some conference members to dial out to a prompted number.

Currently my implementation binds * in conference.xml to the "transfer"
action, leading to a dial plan extension that runs an async full outgoing
event socket. My application serving the event socket does the following:

1) Verifies the user is allowed to dial out, and writes a record to the
database.
2) Retrieves the number to be dialed with 'play_and_get_digits'
3) Bridges the call to the dialed number
4) Waits for the a-leg to press star
5) Uses 'transfer' to transfer both legs back to the conference

Everything works up to step #5, which does nothing.

There is no NOTICE outputted to the console indicating that a transfer is
taking place and neither leg is rejoined to the conference.

I get a normal event back after the transfer sendmsg:

Event-Name: SOCKET_DATA
Content-Type: command/reply
Reply-Text: %2BOK

If I subscribe to all events before running the transfer and receive events
in a loop, I receive nothing but routine RE_SCHEDULE and CALL_UPDATE events
after the transfer.

I am using ESL to communicate with FreeSWITCH, and prior to running any of
steps #1-5 I set the event lock to true. The line from the dial plan
extension that calls my application follows:

<action application="socket" data="172.18.11.1:8001 async full" />

Any ideas? Perhaps I'm "doing it wrong"? I am very new to FS.

Thanks!
Joe Crivello
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From Joe at vsutech.net  Mon May  3 07:47:19 2010
From: Joe at vsutech.net (Joe Crivello [VT])
Date: Mon, 3 May 2010 09:47:19 -0500
Subject: [Freeswitch-users] Trouble With 'transfer' DP and Async Event Socket
Message-ID: <1026F8CF65C4374C820F4FCB50EAA9BE79639671DF@EXC-P-CMS.corp.farwellnet.net>

Hello list,

I am currently working on a conference bridge based on FreeSWITCH that
allows some conference members to dial out to a prompted number.

Currently my implementation binds * in conference.xml to the "transfer"
action, leading to a dial plan extension that runs an async full outgoing
event socket. My application serving the event socket does the following:

1) Verifies the user is allowed to dial out, and writes a record to the
database.
2) Retrieves the number to be dialed with 'play_and_get_digits'
3) Bridges the call to the dialed number
4) Waits for the a-leg to press star
5) Uses 'transfer' to transfer both legs back to the conference

Everything works up to step #5, which does nothing.

There is no NOTICE outputted to the console indicating that a transfer is
taking place and neither leg is rejoined to the conference.

I get a normal event back after the transfer sendmsg:

Event-Name: SOCKET_DATA
Content-Type: command/reply
Reply-Text: %2BOK

If I subscribe to all events before running the transfer and receive events
in a loop, I receive nothing but routine RE_SCHEDULE and CALL_UPDATE events
after the transfer.

I am using ESL to communicate with FreeSWITCH, and prior to running any of
steps #1-5 I set the event lock to true. The line from the dial plan
extension that calls my application follows:

<action application="socket" data="172.18.11.1:8001 async full" />

Any ideas? Perhaps I'm "doing it wrong"? I am very new to FS.

Thanks!
Joe Crivello
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From Joe at vsutech.net  Mon May  3 08:40:53 2010
From: Joe at vsutech.net (Joe Crivello [VT])
Date: Mon, 3 May 2010 10:40:53 -0500
Subject: [Freeswitch-users] Trouble With 'transfer' DP and Async Event Socket
Message-ID: <1026F8CF65C4374C820F4FCB50EAA9BE79639671E1@EXC-P-CMS.corp.farwellnet.net>

Hello list,



I am currently working on a conference bridge based on FreeSWITCH that

allows some conference members to dial out to a prompted number.



Currently my implementation binds * in conference.xml to the "transfer"

action, leading to a dial plan extension that runs an async full outgoing

event socket. My application serving the event socket does the following:



1) Verifies the user is allowed to dial out, and writes a record to the

database.

2) Retrieves the number to be dialed with 'play_and_get_digits'

3) Bridges the call to the dialed number

4) Waits for the a-leg to press star

5) Uses 'transfer' to transfer both legs back to the conference



Everything works up to step #5, which does nothing.



There is no NOTICE outputted to the console indicating that a transfer is

taking place and neither leg is rejoined to the conference.



I get a normal event back after the transfer sendmsg:



Event-Name: SOCKET_DATA

Content-Type: command/reply

Reply-Text: %2BOK



If I subscribe to all events before running the transfer and receive events

in a loop, I receive nothing but routine RE_SCHEDULE and CALL_UPDATE events

after the transfer.



I am using ESL to communicate with FreeSWITCH, and prior to running any of

steps #1-5 I set the event lock to true. The line from the dial plan

extension that calls my application follows:



<action application="socket" data="172.18.11.1:8001 async full" />



Any ideas? Perhaps I'm "doing it wrong"? I am very new to FS.



Thanks!

Joe Crivello



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From david.ponzone at gmail.com  Mon May  3 08:54:23 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Mon, 3 May 2010 17:54:23 +0200
Subject: [Freeswitch-users] Registration ODBC feeded by another
	registrar proxy
In-Reply-To: <4BDED22F.20205@laposte.net>
References: <4BDED22F.20205@laposte.net>
Message-ID: <C5D5A056-B21C-4343-AA13-95D67B3EADA6@gmail.com>

Rod,

Registering every 60 seconds is a bad idea, and this should not be  
justified.
You should register every 1800 seconds and send a NAT keepalive every  
X seconds.
X should be slightly lower than the NAT UDP timeout of the router in  
front of the phones, if the phones are behind NAT.
If the phones are not behind NAT, NAT keepalive is not necessary.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 03/05/2010 ? 15:39, Rod. a ?crit :

> Hi list,
>
> was playing with FS 1.0.6 and trying to test the registration
> performance of FS. (Yes I know FS is more suited as a B2BUA, but  
> please
> read further :p)
>
> So I did the following:
>     - generate one xml file with 20 000 user account like this:
> <include>
> <user id="1">
> <params>
> <param name="password" value="1234"/>
> </params>
> </user>
> <user id="2">
> <params>
> <param name="password" value="1234"/>
> </params>
>             ...
>
> Then I used Sipp to test how many registration per second could be  
> fired
> to the server (quad core 2.83Ghz).
> I setup ulimit variables, and disable nat.
>
> I got this:
>     - using SQL Lite: unable to get higher than 80 registrations per
> second (in fact it's less than this number but didn't test too much  
> this
> setup), I see a lot of retransmission in Sipp
>     - using SQL Lite in ramdisk (tmpfs): OK with 80 registrations per
> second but not much
>     - using ODBC and mysql: 130 reg/sec is OK
>
> With ODBC, above 150 reg/sec I see that FS is stalled to 100-110%  
> CPU, I
> think it's because I'm using only one SIP profile and that SOFIA is
> monothreaded for this SIP profile.
> If I'd like to register every 60sec, the server has to support at  
> least
> more than 300 registration per second.
>
> So I'm wondering if I could setup something like this:
>     - use another SIP Proxy as a registrar and feed the ODBC
> "sip_registration database" of FS
>     - FS will be able to use this database to setup a call
>     - use FS as the outbound proxy for call routing
>
> But what about the user params that have been setup in the xml file
> above. I think that FS loads the user params each time a user is  
> registered.
>
> Comments and advices are welcome.
>
> regards,
> rod.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Mon May  3 08:55:27 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 3 May 2010 10:55:27 -0500
Subject: [Freeswitch-users] Trouble With Transfer and Async Event Socket
In-Reply-To: <n2sf1b753861005030733u73a51a2fh4fae23812f5089@mail.gmail.com>
References: <n2sf1b753861005030733u73a51a2fh4fae23812f5089@mail.gmail.com>
Message-ID: <44D2A064-A5F0-40FB-93ED-56D5DCD480EB@freeswitch.org>

This is your FIRST and LAST warming.. CHILL OUT.

We moderate first post (due to trolls and spammers.).... You have demonstrated you're so impatient that you can't wait a few moments to give someone a chance to tend to the moderation queue.  You posted the same message 5 times from two address... Please do not do this anymore and CHILL OUT... if you needed help that urgent you should have joined IRC.

/b

On May 3, 2010, at 9:33 AM, Joe Crivello wrote:

CONTENT REMOVED

From yehavi.bourvine at gmail.com  Mon May  3 09:01:02 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Mon, 3 May 2010 19:01:02 +0300
Subject: [Freeswitch-users] How to send "waiting call" tone to remote side
Message-ID: <k2zd361ca7b1005030901ta0f4bf0dh4b63df42c2c58ce1@mail.gmail.com>

Hello,

  At the default dialplan of freeswitch, when there is a waiting call the
remote side hears a normal "ring" tone. I would like to change it to send a
"call waiting" ring tone.
What is the official way of doing so? I can think of two ways:

- sending 182 instead of 180.
- send 180 with early media and play the ring tone.

I've tried the second method. It works inside our PBX and at the adjacent
telco PBX. On other telco PBXes it doesn't work (silence). Before I call the
telco I would like to know what is the common way of doing so.

                           Thanks! __yehavi:
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From brian at freeswitch.org  Mon May  3 09:08:00 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 3 May 2010 11:08:00 -0500
Subject: [Freeswitch-users] How to send "waiting call" tone to remote
	side
In-Reply-To: <k2zd361ca7b1005030901ta0f4bf0dh4b63df42c2c58ce1@mail.gmail.com>
References: <k2zd361ca7b1005030901ta0f4bf0dh4b63df42c2c58ce1@mail.gmail.com>
Message-ID: <E203337E-48EF-470E-9DFC-4C6CA9E0FFCF@freeswitch.org>

Not our job.. thats the endpoints job.

/b

On May 3, 2010, at 11:01 AM, Yehavi Bourvine wrote:

> Hello,
>  
>   At the default dialplan of freeswitch, when there is a waiting call the remote side hears a normal "ring" tone. I would like to change it to send a "call waiting" ring tone.
> What is the official way of doing so? I can think of two ways:
>  
> - sending 182 instead of 180.
> - send 180 with early media and play the ring tone.
>  
> I've tried the second method. It works inside our PBX and at the adjacent telco PBX. On other telco PBXes it doesn't work (silence). Before I call the telco I would like to know what is the common way of doing so.
>  
>                            Thanks! __yehavi:
>  



From yehavi.bourvine at gmail.com  Mon May  3 09:17:33 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Mon, 3 May 2010 19:17:33 +0300
Subject: [Freeswitch-users] How to send "waiting call" tone to remote
	side
In-Reply-To: <E203337E-48EF-470E-9DFC-4C6CA9E0FFCF@freeswitch.org>
References: <k2zd361ca7b1005030901ta0f4bf0dh4b63df42c2c58ce1@mail.gmail.com>
	<E203337E-48EF-470E-9DFC-4C6CA9E0FFCF@freeswitch.org>
Message-ID: <j2sd361ca7b1005030917sa4c0ccf9z4f0e543a16762526@mail.gmail.com>

The endpoint (Polycom, SNOM) doesn't do it, so I have to do it...

                   Thanks, __Yehavi:

2010/5/3 Brian West <brian at freeswitch.org>

> Not our job.. thats the endpoints job.
>
> /b
>
> On May 3, 2010, at 11:01 AM, Yehavi Bourvine wrote:
>
> > Hello,
> >
> >   At the default dialplan of freeswitch, when there is a waiting call the
> remote side hears a normal "ring" tone. I would like to change it to send a
> "call waiting" ring tone.
> > What is the official way of doing so? I can think of two ways:
> >
> > - sending 182 instead of 180.
> > - send 180 with early media and play the ring tone.
> >
> > I've tried the second method. It works inside our PBX and at the adjacent
> telco PBX. On other telco PBXes it doesn't work (silence). Before I call the
> telco I would like to know what is the common way of doing so.
> >
> >                            Thanks! __yehavi:
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From kward at binarysignal.com  Mon May  3 09:30:55 2010
From: kward at binarysignal.com (Kurt Ward)
Date: Mon, 3 May 2010 09:30:55 -0700
Subject: [Freeswitch-users] External SIP profile
Message-ID: <56DF79F1-2709-49CC-B5A3-C3A733F2B26E@binarysignal.com>

Maybe this is a bit of an uneducated noob question (or  
misunderstanding about SIP) but here goes:

If I have an external profile set up and registering, does it  
essentially behave like a soft phone would behave?

What I am running into is incoming calls work fine, but outgoing calls  
always produce a 404.  If I use a soft phone using the same  
registration I can make calls with no problem (in both scenarios the  
calls are to/from 4 digit extensions on a Mitel PBX). 
  


From Joe at vsutech.net  Mon May  3 09:43:04 2010
From: Joe at vsutech.net (Joe Crivello [VT])
Date: Mon, 3 May 2010 11:43:04 -0500
Subject: [Freeswitch-users] Trouble With Transfer and Async Event Socket
In-Reply-To: <44D2A064-A5F0-40FB-93ED-56D5DCD480EB@freeswitch.org>
References: <n2sf1b753861005030733u73a51a2fh4fae23812f5089@mail.gmail.com>
	<44D2A064-A5F0-40FB-93ED-56D5DCD480EB@freeswitch.org>
Message-ID: <1026F8CF65C4374C820F4FCB50EAA9BE79639671EE@EXC-P-CMS.corp.farwellnet.net>

Brian,

I had no intention of spamming the list. I apologize.

I sent the original message early this morning and it did not appear in the
archive or send me a receipt for the email (when messages from other folks
were going through).

I asked a few people in the IRC chat room and the consensus was that there
was a problem with my email server. I had no idea that the list was
moderated (and I did look for documentation of such).

I simply thought my email server was misbehaving.

Again, my apologies.

-Joe

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Monday, May 03, 2010 10:55 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Trouble With Transfer and Async Event Socket

This is your FIRST and LAST warming.. CHILL OUT.

We moderate first post (due to trolls and spammers.).... You have
demonstrated you're so impatient that you can't wait a few moments to give
someone a chance to tend to the moderation queue.  You posted the same
message 5 times from two address... Please do not do this anymore and CHILL
OUT... if you needed help that urgent you should have joined IRC.

/b

On May 3, 2010, at 9:33 AM, Joe Crivello wrote:

CONTENT REMOVED
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From brian at freeswitch.org  Mon May  3 09:48:01 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 3 May 2010 11:48:01 -0500
Subject: [Freeswitch-users] Trouble With Transfer and Async Event Socket
In-Reply-To: <1026F8CF65C4374C820F4FCB50EAA9BE79639671EE@EXC-P-CMS.corp.farwellnet.net>
References: <n2sf1b753861005030733u73a51a2fh4fae23812f5089@mail.gmail.com>
	<44D2A064-A5F0-40FB-93ED-56D5DCD480EB@freeswitch.org>
	<1026F8CF65C4374C820F4FCB50EAA9BE79639671EE@EXC-P-CMS.corp.farwellnet.net>
Message-ID: <83E48B85-264E-4BA8-B00D-B6252F2EBB89@freeswitch.org>

Forgiven this time.  ;)  Welcome to FreeSWITCH.

/b

On May 3, 2010, at 11:43 AM, Joe Crivello [VT] wrote:

> Brian,
> 
> I had no intention of spamming the list. I apologize.
> 
> I sent the original message early this morning and it did not appear in the
> archive or send me a receipt for the email (when messages from other folks
> were going through).
> 
> I asked a few people in the IRC chat room and the consensus was that there
> was a problem with my email server. I had no idea that the list was
> moderated (and I did look for documentation of such).
> 
> I simply thought my email server was misbehaving.
> 
> Again, my apologies.
> 
> -Joe



From kawarod at laposte.net  Mon May  3 10:06:54 2010
From: kawarod at laposte.net (Rod.)
Date: Mon, 03 May 2010 21:06:54 +0400
Subject: [Freeswitch-users] Registration ODBC feeded by
 another	registrar proxy
In-Reply-To: <C5D5A056-B21C-4343-AA13-95D67B3EADA6@gmail.com>
References: <4BDED22F.20205@laposte.net>
	<C5D5A056-B21C-4343-AA13-95D67B3EADA6@gmail.com>
Message-ID: <4BDF02AE.3010003@laposte.net>

Hi,

thanks for your answer and just some details to describe what I'm 
looking for.
I have to register 25 000 subscribers, no NAT is involved, each 
equipment has its own IP address.
These equipments are registering every 60 seconds on our current 
platform, but I can change this parameter if needed.
Equipments are ADSL CPE (router), that's why I'm using 60sec cause 
flapping could happen very often with ADSL if the copper line is crappy. 
ADSL could be very unpredictable sometimes.
As I don't want to delay too much forwarding to voicemail if a user is 
unavailable (network issue), 60 sec was chosen (bandwith is not an 
issue). But as I told before, I'm open to your suggestions.

To Philip, using a single SIP proxy (opensips/ser...) in front of a FS 
cluster could be a single point of failure too.
I think that maybe a solution using DNS SRV to distribute the load 
across a cluster could do the trick or some kind of LVS (virtual IP 
shared across many servers)
XML curl is a good idea too.
To be honest, clustering is a must to avoid a single point of failure, 
but FS performance as a SBC are really great even on commodity hardware, 
more than 100 CallPerSecond with no transcoding. That's why I think that 
a mix with a SIP registrar and FS (and redundancy) could easily handle 
my 25 000 subscribers. I did some lab (one or 2 years ago) with Kamailio 
registering 90 000 users every 60sec (1500 Registration per second) 
without any issues.
In my network, 25 000 users are not pushing more than 10 CPS and 500 
simultaneous call. I'm not doing VoIP termination.

At the moment, I'm just collecting data/feedback on what could be done 
as I have some time to work on this project, and if going further I will 
share the configuration as I did before:
http://wiki.freeswitch.org/wiki/SBC_Setup (not the best setup, but hope 
it helps users to begin with FS)

regards,
rod.




Le 03/05/2010 19:54, David Ponzone a ?crit :
> Rod,
>
> Registering every 60 seconds is a bad idea, and this should not be 
> justified.
> You should register every 1800 seconds and send a NAT keepalive every 
> X seconds.
> X should be slightly lower than the NAT UDP timeout of the router in 
> front of the phones, if the phones are behind NAT.
> If the phones are not behind NAT, NAT keepalive is not necessary.
>
> David Ponzone Direction Technique
> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - www.ipeva-studio.com 
> <BLOCKED::http://www.ipeva-studio.com/>
>
> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis 
> ? l'intention exclusive de ses destinataires. Toute utilisation ou 
> diffusion non autoris?e est interdite. Tout message ?lectronique est 
> susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au 
> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous 
> n'?tes pas destinataire de ce message, merci de le d?truire 
> imm?diatement et d'avertir l'exp?diteur./
> /
> /
>
>
>
> Le 03/05/2010 ? 15:39, Rod. a ?crit :
>
>> Hi list,
>>
>> was playing with FS 1.0.6 and trying to test the registration
>> performance of FS. (Yes I know FS is more suited as a B2BUA, but please
>> read further :p)
>>
>> So I did the following:
>>     - generate one xml file with 20 000 user account like this:
>> <include>
>> <user id="1">
>> <params>
>> <param name="password" value="1234"/>
>> </params>
>> </user>
>> <user id="2">
>> <params>
>> <param name="password" value="1234"/>
>> </params>
>>             ...
>>
>> Then I used Sipp to test how many registration per second could be fired
>> to the server (quad core 2.83Ghz).
>> I setup ulimit variables, and disable nat.
>>
>> I got this:
>>     - using SQL Lite: unable to get higher than 80 registrations per
>> second (in fact it's less than this number but didn't test too much this
>> setup), I see a lot of retransmission in Sipp
>>     - using SQL Lite in ramdisk (tmpfs): OK with 80 registrations per
>> second but not much
>>     - using ODBC and mysql: 130 reg/sec is OK
>>
>> With ODBC, above 150 reg/sec I see that FS is stalled to 100-110% CPU, I
>> think it's because I'm using only one SIP profile and that SOFIA is
>> monothreaded for this SIP profile.
>> If I'd like to register every 60sec, the server has to support at least
>> more than 300 registration per second.
>>
>> So I'm wondering if I could setup something like this:
>>     - use another SIP Proxy as a registrar and feed the ODBC
>> "sip_registration database" of FS
>>     - FS will be able to use this database to setup a call
>>     - use FS as the outbound proxy for call routing
>>
>> But what about the user params that have been setup in the xml file
>> above. I think that FS loads the user params each time a user is 
>> registered.
>>
>> Comments and advices are welcome.
>>
>> regards,
>> rod.
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org 
>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    

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From janvb at live.com  Mon May  3 10:42:30 2010
From: janvb at live.com (Jan Berger)
Date: Mon, 3 May 2010 19:42:30 +0200
Subject: [Freeswitch-users] Registration ODBC feeded by
	another	registrar proxy
In-Reply-To: <4BDF02AE.3010003@laposte.net>
References: <4BDED22F.20205@laposte.net>,
	<C5D5A056-B21C-4343-AA13-95D67B3EADA6@gmail.com>,
	<4BDF02AE.3010003@laposte.net>
Message-ID: <SNT135-w18CD702457BE5E08B7D51D2F20@phx.gbl>


May a suggest a change filter developed if this really is needed?

 

Re-loading everything just in case something has changes is a huge waste of bandwidth and CPU - if you install an intelligent change filter you would be down to a few entries changing.

 

Jan
 


Date: Mon, 3 May 2010 21:06:54 +0400
From: kawarod at laposte.net
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Registration ODBC feeded by another registrar proxy

Hi,

thanks for your answer and just some details to describe what I'm looking for.
I have to register 25 000 subscribers, no NAT is involved, each equipment has its own IP address.
These equipments are registering every 60 seconds on our current platform, but I can change this parameter if needed.
Equipments are ADSL CPE (router), that's why I'm using 60sec cause flapping could happen very often with ADSL if the copper line is crappy. ADSL could be very unpredictable sometimes.
As I don't want to delay too much forwarding to voicemail if a user is unavailable (network issue), 60 sec was chosen (bandwith is not an issue). But as I told before, I'm open to your suggestions.

To Philip, using a single SIP proxy (opensips/ser...) in front of a FS cluster could be a single point of failure too.
I think that maybe a solution using DNS SRV to distribute the load across a cluster could do the trick or some kind of LVS (virtual IP shared across many servers)
XML curl is a good idea too.
To be honest, clustering is a must to avoid a single point of failure, but FS performance as a SBC are really great even on commodity hardware, more than 100 CallPerSecond with no transcoding. That's why I think that a mix with a SIP registrar and FS (and redundancy) could easily handle my 25 000 subscribers. I did some lab (one or 2 years ago) with Kamailio registering 90 000 users every 60sec (1500 Registration per second) without any issues.
In my network, 25 000 users are not pushing more than 10 CPS and 500 simultaneous call. I'm not doing VoIP termination.

At the moment, I'm just collecting data/feedback on what could be done as I have some time to work on this project, and if going further I will share the configuration as I did before:
http://wiki.freeswitch.org/wiki/SBC_Setup (not the best setup, but hope it helps users to begin with FS)

regards,
rod.




Le 03/05/2010 19:54, David Ponzone a ?crit : 

Rod,

Registering every 60 seconds is a bad idea, and this should not be justified. 
You should register every 1800 seconds and send a NAT keepalive every X seconds.
X should be slightly lower than the NAT UDP timeout of the router in front of the phones, if the phones are behind NAT.
If the phones are not behind NAT, NAT keepalive is not necessary.





David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34


Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr  -   www.ipeva-studio.com



Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 03/05/2010 ? 15:39, Rod. a ?crit :

Hi list,

was playing with FS 1.0.6 and trying to test the registration 
performance of FS. (Yes I know FS is more suited as a B2BUA, but please 
read further :p)

So I did the following:
    - generate one xml file with 20 000 user account like this:
<include>
<user id="1">
<params>
<param name="password" value="1234"/>
</params>
</user>
<user id="2">
<params>
<param name="password" value="1234"/>
</params>
            ...

Then I used Sipp to test how many registration per second could be fired 
to the server (quad core 2.83Ghz).
I setup ulimit variables, and disable nat.

I got this:
    - using SQL Lite: unable to get higher than 80 registrations per 
second (in fact it's less than this number but didn't test too much this 
setup), I see a lot of retransmission in Sipp
    - using SQL Lite in ramdisk (tmpfs): OK with 80 registrations per 
second but not much
    - using ODBC and mysql: 130 reg/sec is OK

With ODBC, above 150 reg/sec I see that FS is stalled to 100-110% CPU, I 
think it's because I'm using only one SIP profile and that SOFIA is 
monothreaded for this SIP profile.
If I'd like to register every 60sec, the server has to support at least 
more than 300 registration per second.

So I'm wondering if I could setup something like this:
    - use another SIP Proxy as a registrar and feed the ODBC 
"sip_registration database" of FS
    - FS will be able to use this database to setup a call
    - use FS as the outbound proxy for call routing

But what about the user params that have been setup in the xml file 
above. I think that FS loads the user params each time a user is registered.

Comments and advices are welcome.

regards,
rod.



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From kenfulmer at icstechnologysolutions.com  Mon May  3 10:44:20 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Mon, 3 May 2010 12:44:20 -0500
Subject: [Freeswitch-users] absolute_codec_string vs codec_string
Message-ID: <010301caeae8$42dc8160$c8958420$@com>

Would someone please clarify the difference between the following
parameters? I've seen the Wiki page but still don't understand the
difference. 

 

1.       absolute_codec_string

 

2.       codec_string

 

Thanks,

 

Ken Fulmer

 

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From anthony.minessale at gmail.com  Mon May  3 11:03:18 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 3 May 2010 13:03:18 -0500
Subject: [Freeswitch-users] absolute_codec_string vs codec_string
In-Reply-To: <010301caeae8$42dc8160$c8958420$@com>
References: <010301caeae8$42dc8160$c8958420$@com>
Message-ID: <z2s191c3a031005031103m1996db49kc476e85a8a2f4d7@mail.gmail.com>

codec_string is a list of codecs you would like to advertise, in addition if
the outgoing call is being originated from an inbound call the codec of that
inbound call will be prepeneded to the list to avoid transcoding.

absolute_codec_string is an exact list of codecs you want to offer and it
will not add any extra ones automatically.


On Mon, May 3, 2010 at 12:44 PM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  Would someone please clarify the difference between the following
> parameters? I?ve seen the Wiki page but still don?t understand the
> difference.
>
>
>
> 1.       absolute_codec_string
>
>
>
> 2.       codec_string
>
>
>
> Thanks,
>
>
>
> Ken Fulmer
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <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
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From anthony.minessale at gmail.com  Mon May  3 11:05:47 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 3 May 2010 13:05:47 -0500
Subject: [Freeswitch-users] Trouble With Transfer and Async Event Socket
In-Reply-To: <n2sf1b753861005030733u73a51a2fh4fae23812f5089@mail.gmail.com>
References: <n2sf1b753861005030733u73a51a2fh4fae23812f5089@mail.gmail.com>
Message-ID: <n2y191c3a031005031105v4cd7ad3by96e59cbfc19b319@mail.gmail.com>

What command are you using to try to transfer it to your conference?


On Mon, May 3, 2010 at 9:33 AM, Joe Crivello <josephcrivello at gmail.com>wrote:

> Hello All,
>
> I am currently working on a conference bridge based on FreeSWITCH that
> allows some conference members to dial out to a prompted number.
>
> Currently my implementation binds * in conference.xml to the "transfer"
> action, leading to a dial plan extension that runs an async full outgoing
> event socket. My application serving the event socket does the following:
>
> 1) Verifies the user is allowed to dial out, and writes a record to the
> database.
> 2) Retrieves the number to be dialed with 'play_and_get_digits'
> 3) Bridges the call to the dialed number
> 4) Waits for the a-leg to press star
> 5) Uses 'transfer' to transfer both legs back to the conference
>
> Everything works up to step #5, which does nothing.
>
> There is no NOTICE outputted to the console indicating that a transfer is
> taking place and neither leg is rejoined to the conference.
>
> I get a normal event back after the transfer sendmsg:
>
> Event-Name: SOCKET_DATA
> Content-Type: command/reply
> Reply-Text: %2BOK
>
> If I subscribe to all events before running the transfer and receive events
> in a loop, I receive nothing but routine RE_SCHEDULE and CALL_UPDATE events
> after the transfer.
>
> I am using ESL to communicate with FreeSWITCH, and prior to running any of
> steps #1-5 I set the event lock to true. The line from the dial plan
> extension that calls my application follows:
>
> <action application="socket" data="172.18.11.1:8001 async full" />
>
> Any ideas? Perhaps I'm "doing it wrong"? I am very new to FS.
>
> Thanks!
> Joe Crivello
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <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
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From vfclists at googlemail.com  Mon May  3 11:05:41 2010
From: vfclists at googlemail.com (Frank Church)
Date: Mon, 3 May 2010 19:05:41 +0100
Subject: [Freeswitch-users] How to monitor find out which events are
	currently monitored, set 	only a particular group of events?
Message-ID: <k2w7abab2411005031105n726c11dx434981234cce49f8@mail.gmail.com>

Is there a way of getting of list the event types that are being monitored
by your connection to Freeswitch, other than those keeping a record of those
you've added from your own end?

I want to add some without deleting what is present with the first event
plain XXXX or  filter Event-Name XXXX command

-- 
Frank Church

=======================
http://devblog.brahmancreations.com
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From aep.lists at it46.se  Mon May  3 11:19:07 2010
From: aep.lists at it46.se (Alberto Escudero)
Date: Mon, 3 May 2010 20:19:07 +0200
Subject: [Freeswitch-users] wrong ptimes and second SDP!,
	cough cough Thomson ST2030
Message-ID: <b57c607101b188410fac1458fd0f4edf.squirrel@correo.nodo50.org>

I  have a Thomson ST2030 VoIP phone that seams to be sending wrong ptimes.
The SDP says 20 ms when it should be sending 10 ms.

Freeswitch *cough cough* algorithm is able to set the soft timer for 10
ms... but when another SDP arrives (after approx. 30-40 seconds) the ptime
is set back again to 20 ms.

I am using 1.0.5 and it seems a bug in Thomson SIP stack.

Any work around?
-- 
Stopping junk mailers is good for the environment





From josephcrivello at gmail.com  Mon May  3 11:20:14 2010
From: josephcrivello at gmail.com (Joseph Crivello)
Date: Mon, 3 May 2010 13:20:14 -0500
Subject: [Freeswitch-users] Trouble With Transfer and Async Event Socket
In-Reply-To: <n2y191c3a031005031105v4cd7ad3by96e59cbfc19b319@mail.gmail.com>
References: <n2sf1b753861005030733u73a51a2fh4fae23812f5089@mail.gmail.com>
	<n2y191c3a031005031105v4cd7ad3by96e59cbfc19b319@mail.gmail.com>
Message-ID: <012d01caeaed$47fc3460$d7f49d20$@com>

Example command:

 

Command: transfer

Arguments: -both ConfXfer-1 XML default

 

And here is the referenced dialplan extension:

 

    <extension name="ConfXfer">

      <condition field="destination_number" expression="^ConfXfer-(\d+)$">

        <action application="answer"/>

        <action application="conference" data="$1" />

      </condition>

    </extension>

 

Interestingly, I recently discovered that the transfer works if I do it
before the bridge finishes (figured that out by accident).

 

-Joe

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, May 03, 2010 1:06 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Trouble With Transfer and Async Event Socket

 

What command are you using to try to transfer it to your conference?

 

On Mon, May 3, 2010 at 9:33 AM, Joe Crivello <josephcrivello at gmail.com>
wrote:

Hello All,

I am currently working on a conference bridge based on FreeSWITCH that
allows some conference members to dial out to a prompted number.

Currently my implementation binds * in conference.xml to the "transfer"
action, leading to a dial plan extension that runs an async full outgoing
event socket. My application serving the event socket does the following:

1) Verifies the user is allowed to dial out, and writes a record to the
database.
2) Retrieves the number to be dialed with 'play_and_get_digits'
3) Bridges the call to the dialed number
4) Waits for the a-leg to press star
5) Uses 'transfer' to transfer both legs back to the conference

Everything works up to step #5, which does nothing.

There is no NOTICE outputted to the console indicating that a transfer is
taking place and neither leg is rejoined to the conference.

I get a normal event back after the transfer sendmsg:

Event-Name: SOCKET_DATA
Content-Type: command/reply
Reply-Text: %2BOK

If I subscribe to all events before running the transfer and receive events
in a loop, I receive nothing but routine RE_SCHEDULE and CALL_UPDATE events
after the transfer.

I am using ESL to communicate with FreeSWITCH, and prior to running any of
steps #1-5 I set the event lock to true. The line from the dial plan
extension that calls my application follows:

<action application="socket" data=" <http://172.18.11.1:8001/>
172.18.11.1:8001 async full" />

Any ideas? Perhaps I'm "doing it wrong"? I am very new to FS.

Thanks!
Joe Crivello 
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
Anthony Minessale II

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

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

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

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From brian at freeswitch.org  Mon May  3 11:28:09 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 3 May 2010 13:28:09 -0500
Subject: [Freeswitch-users] wrong ptimes and second SDP!,
	cough cough Thomson ST2030
In-Reply-To: <b57c607101b188410fac1458fd0f4edf.squirrel@correo.nodo50.org>
References: <b57c607101b188410fac1458fd0f4edf.squirrel@correo.nodo50.org>
Message-ID: <31AF5567-8AE7-4ADD-A611-804C6BC6C58E@freeswitch.org>

Just set codec negotiation to scrooge.
/b

On May 3, 2010, at 1:19 PM, Alberto Escudero wrote:

> I  have a Thomson ST2030 VoIP phone that seams to be sending wrong ptimes.
> The SDP says 20 ms when it should be sending 10 ms.
> 
> Freeswitch *cough cough* algorithm is able to set the soft timer for 10
> ms... but when another SDP arrives (after approx. 30-40 seconds) the ptime
> is set back again to 20 ms.
> 
> I am using 1.0.5 and it seems a bug in Thomson SIP stack.
> 
> Any work around?
> -- 
> Stopping junk mailers is good for the environment



From oseslija at gmail.com  Mon May  3 11:38:57 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Mon, 3 May 2010 20:38:57 +0200
Subject: [Freeswitch-users] How to send "waiting call" tone to remote
	side
In-Reply-To: <j2sd361ca7b1005030917sa4c0ccf9z4f0e543a16762526@mail.gmail.com>
References: <k2zd361ca7b1005030901ta0f4bf0dh4b63df42c2c58ce1@mail.gmail.com>
	<E203337E-48EF-470E-9DFC-4C6CA9E0FFCF@freeswitch.org>
	<j2sd361ca7b1005030917sa4c0ccf9z4f0e543a16762526@mail.gmail.com>
Message-ID: <x2n4468a6771005031138wefe21240hbbe316ff4dd62b3f@mail.gmail.com>

Polycom and snom can do cw.
O.

On May 3, 2010 6:23 PM, "Yehavi Bourvine" <yehavi.bourvine at gmail.com> wrote:

The endpoint (Polycom, SNOM) doesn't do it, so I have to do it...

                   Thanks, __Yehavi:

2010/5/3 Brian West <brian at freeswitch.org>


>
> Not our job.. thats the endpoints job.
>
> /b
>
> On May 3, 2010, at 11:01 AM, Yehavi Bourvine ...

_______________________________________________
FreeSWITCH-users mailing list
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From kenfulmer at icstechnologysolutions.com  Mon May  3 11:40:01 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Mon, 3 May 2010 13:40:01 -0500
Subject: [Freeswitch-users] absolute_codec_string vs codec_string
In-Reply-To: <z2s191c3a031005031103m1996db49kc476e85a8a2f4d7@mail.gmail.com>
References: <010301caeae8$42dc8160$c8958420$@com>
	<z2s191c3a031005031103m1996db49kc476e85a8a2f4d7@mail.gmail.com>
Message-ID: <013801caeaf0$0a8772e0$1f9658a0$@com>

Thanks!

 

Ken

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, May 03, 2010 1:03 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] absolute_codec_string vs codec_string

 

codec_string is a list of codecs you would like to advertise, in addition if
the outgoing call is being originated from an inbound call the codec of that
inbound call will be prepeneded to the list to avoid transcoding.

 

absolute_codec_string is an exact list of codecs you want to offer and it
will not add any extra ones automatically. 

 

On Mon, May 3, 2010 at 12:44 PM, Ken Fulmer
<kenfulmer at icstechnologysolutions.com> wrote:

Would someone please clarify the difference between the following
parameters? I've seen the Wiki page but still don't understand the
difference. 

 

1.       absolute_codec_string

 

2.       codec_string

 

Thanks,

 

Ken Fulmer

 


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




-- 
Anthony Minessale II

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

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

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

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From robert.hadley at teotech.com  Mon May  3 11:52:24 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Mon, 3 May 2010 11:52:24 -0700
Subject: [Freeswitch-users] Anyone have analog Fax working through FS and
	Sangoma cards?
Message-ID: <83B388ED98C64D23A32EBC8828A68E19@greyhawk.tonecommander.com>

 

Does anyone have analog Fax working through Freeswitch and Sangoma A101 &
A200/FXS cards?  If so, would you share what configuration and dialplan
settings need to be made?  I have a problem receiving Faxes, which only work
80% of the time and often take 10-12 minutes per Fax, versus 2 minutes when
using the Fax machine on analog POTS line.

 

My setup: SendingFax --- {PSTN} --- PRI --- A101 --- FS --- A200/FXS ---
MyFax

 

 

Pastebin to dialplan, wanpipe conf, vars.xml, internal.xml:
http://pastebin.freeswitch.org/12884 

 

 

Any help would be greatly appreciated.

 

Thanks,

Robert

 

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From yehavi.bourvine at gmail.com  Mon May  3 11:53:54 2010
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Mon, 3 May 2010 21:53:54 +0300
Subject: [Freeswitch-users] How to send "waiting call" tone to remote
	side
In-Reply-To: <x2n4468a6771005031138wefe21240hbbe316ff4dd62b3f@mail.gmail.com>
References: <k2zd361ca7b1005030901ta0f4bf0dh4b63df42c2c58ce1@mail.gmail.com>
	<E203337E-48EF-470E-9DFC-4C6CA9E0FFCF@freeswitch.org>
	<j2sd361ca7b1005030917sa4c0ccf9z4f0e543a16762526@mail.gmail.com>
	<x2n4468a6771005031138wefe21240hbbe316ff4dd62b3f@mail.gmail.com>
Message-ID: <y2rd361ca7b1005031153pede715ees5ca5811fb558bc1d@mail.gmail.com>

I think I did not make myself clear.

Suppose A on the public PSTN calls B who is on my Freeswitch, and suppose B
is already on a call. I want A to hear a stuttered "other side is ringing"
call so he knows that B is on another call.

What I would like to know is whether there is some standard way to signal
the PBX of A to send him this stuttered tone.

                         Thanks, __Yehavi:

 2010/5/3 Ognjen Seslija <oseslija at gmail.com>

>  Polycom and snom can do cw.
> O.
>
>  On May 3, 2010 6:23 PM, "Yehavi Bourvine" <yehavi.bourvine at gmail.com>
> wrote:
>
>  The endpoint (Polycom, SNOM) doesn't do it, so I have to do it...
>
>                    Thanks, __Yehavi:
>
>  2010/5/3 Brian West <brian at freeswitch.org>
>
>
> >
> > Not our job.. thats the endpoints job.
> >
> > /b
> >
> > On May 3, 2010, at 11:01 AM, Yehavi Bourvine ...
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
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From aep.lists at it46.se  Mon May  3 11:59:37 2010
From: aep.lists at it46.se (Alberto Escudero)
Date: Mon, 3 May 2010 20:59:37 +0200
Subject: [Freeswitch-users] wrong ptimes and second SDP!,
 cough cough Thomson ST2030
In-Reply-To: <31AF5567-8AE7-4ADD-A611-804C6BC6C58E@freeswitch.org>
References: <b57c607101b188410fac1458fd0f4edf.squirrel@correo.nodo50.org>
	<31AF5567-8AE7-4ADD-A611-804C6BC6C58E@freeswitch.org>
Message-ID: <06bafa16896d6e54cc1d5c4e8e5a44bd.squirrel@correo.nodo50.org>

I set in internal.xml

<param name="inbound-codec-negotiation" value="scrooge"/>

But the second SDP returns the ptime to the wrong time.

See below:
2010-05-03 20:56:38.944288 [DEBUG] sofia_glue.c:3596 Bah HUMBUG! Sticking
with PCMU at 8000h@20i
2010-05-03 20:56:38.944288 [DEBUG] sofia_glue.c:2283 Changing Codec from
PCMU at 10ms to PCMU at 20ms

--

2010-05-03 20:55:48.879237 [DEBUG] switch_core_codec.c:122
sofia/internal/1002 at 192.168.1.12:5060 Push codec L16:10
2010-05-03 20:56:38.930237 [DEBUG] sofia.c:4153 Channel
sofia/internal/1002 at 192.168.1.12:5060 entering state [received][100]
2010-05-03 20:56:38.931244 [DEBUG] sofia.c:4153 Channel
sofia/internal/1002 at 192.168.1.12:5060 entering state [completed][200]
2010-05-03 20:56:38.943283 [DEBUG] sofia.c:4153 Channel
sofia/internal/1002 at 192.168.1.12:5060 entering state [ready][200]
2010-05-03 20:56:38.943283 [DEBUG] sofia.c:4161 Duplicate SDP
v=0
o=SIP-IPPhone-0000 102406833 102406833 IN IP4 192.168.1.65
s=RTP Audio
c=IN IP4 192.168.1.65
t=0 0
m=audio 41000 RTP/AVP 0 97
a=rtpmap:0 PCMU/8000
a=rtpmap:97 telephone-event/8000

2010-05-03 20:56:38.943283 [DEBUG] sofia_glue.c:3585 Audio Codec Compare
[PCMU:0:8000:20]/[G7221:115:32000:20]
2010-05-03 20:56:38.943283 [DEBUG] sofia_glue.c:3524 Set 2833 dtmf
send/recv payload to 97
2010-05-03 20:56:38.943283 [DEBUG] sofia_glue.c:3585 Audio Codec Compare
[telephone-event:97:8000:20]/[G7221:115:32000:20]
2010-05-03 20:56:38.943283 [DEBUG] sofia_glue.c:3585 Audio Codec Compare
[PCMU:0:8000:20]/[G7221:107:16000:20]
2010-05-03 20:56:38.943283 [DEBUG] sofia_glue.c:3585 Audio Codec Compare
[telephone-event:97:8000:20]/[G7221:107:16000:20]
2010-05-03 20:56:38.943283 [DEBUG] sofia_glue.c:3585 Audio Codec Compare
[PCMU:0:8000:20]/[G722:9:8000:20]
2010-05-03 20:56:38.943283 [DEBUG] sofia_glue.c:3585 Audio Codec Compare
[telephone-event:97:8000:20]/[G722:9:8000:20]
2010-05-03 20:56:38.943283 [DEBUG] sofia_glue.c:3585 Audio Codec Compare
[PCMU:0:8000:20]/[PCMU:0:8000:20]
2010-05-03 20:56:38.944288 [DEBUG] sofia_glue.c:3596 Bah HUMBUG! Sticking
with PCMU at 8000h@20i
2010-05-03 20:56:38.944288 [DEBUG] sofia_glue.c:2283 Changing Codec from
PCMU at 10ms to PCMU at 20ms
2010-05-03 20:56:38.947289 [DEBUG] switch_rtp.c:1080 RE-Starting timer
[soft] 160 bytes per 20000ms
2010-05-03 20:56:38.947289 [DEBUG] sofia_glue.c:2354 Set Codec
sofia/internal/1002 at 192.168.1.12:5060 PCMU/8000 20 ms 160 samples
2010-05-03 20:56:38.947289 [DEBUG] switch_core_codec.c:122
sofia/internal/1002 at 192.168.1.12:5060 Push codec PCMU:0
2010-05-03 20:56:38.947289 [DEBUG] sofia_glue.c:2579 Audio params are
unchanged for sofia/internal/1002 at 192.168.1.12:5060.

-- 
Stopping junk mailers is good for the environment

> Just set codec negotiation to scrooge.
> /b
>
> On May 3, 2010, at 1:19 PM, Alberto Escudero wrote:
>
>> I  have a Thomson ST2030 VoIP phone that seams to be sending wrong
>> ptimes.
>> The SDP says 20 ms when it should be sending 10 ms.
>>
>> Freeswitch *cough cough* algorithm is able to set the soft timer for 10
>> ms... but when another SDP arrives (after approx. 30-40 seconds) the
>> ptime
>> is set back again to 20 ms.
>>
>> I am using 1.0.5 and it seems a bug in Thomson SIP stack.
>>
>> Any work around?
>> --
>> Stopping junk mailers is good for the environment
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>




From oseslija at gmail.com  Mon May  3 12:03:01 2010
From: oseslija at gmail.com (Ognjen Seslija)
Date: Mon, 3 May 2010 21:03:01 +0200
Subject: [Freeswitch-users] How to send "waiting call" tone to remote
	side
In-Reply-To: <y2rd361ca7b1005031153pede715ees5ca5811fb558bc1d@mail.gmail.com>
References: <k2zd361ca7b1005030901ta0f4bf0dh4b63df42c2c58ce1@mail.gmail.com>
	<E203337E-48EF-470E-9DFC-4C6CA9E0FFCF@freeswitch.org>
	<j2sd361ca7b1005030917sa4c0ccf9z4f0e543a16762526@mail.gmail.com>
	<x2n4468a6771005031138wefe21240hbbe316ff4dd62b3f@mail.gmail.com>
	<y2rd361ca7b1005031153pede715ees5ca5811fb558bc1d@mail.gmail.com>
Message-ID: <o2o4468a6771005031203tceb64412h3cfefbf944dc0eca@mail.gmail.com>

If the endpoint didn't turn on call waiting you're stuck with (most
probably) SIP 486 message from it (USER_BUSY).
O.



On Mon, May 3, 2010 at 8:53 PM, Yehavi Bourvine
<yehavi.bourvine at gmail.com>wrote:

> I think I did not make myself clear.
>
> Suppose A on the public PSTN calls B who is on my Freeswitch, and suppose B
> is already on a call. I want A to hear a stuttered "other side is ringing"
> call so he knows that B is on another call.
>
> What I would like to know is whether there is some standard way to signal
> the PBX of A to send him this stuttered tone.
>
>                          Thanks, __Yehavi:
>
>  2010/5/3 Ognjen Seslija <oseslija at gmail.com>
>
>>  Polycom and snom can do cw.
>> O.
>>
>>  On May 3, 2010 6:23 PM, "Yehavi Bourvine" <yehavi.bourvine at gmail.com>
>> wrote:
>>
>>  The endpoint (Polycom, SNOM) doesn't do it, so I have to do it...
>>
>>                    Thanks, __Yehavi:
>>
>>  2010/5/3 Brian West <brian at freeswitch.org>
>>
>>
>> >
>> > Not our job.. thats the endpoints job.
>> >
>> > /b
>> >
>> > On May 3, 2010, at 11:01 AM, Yehavi Bourvine ...
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Mon May  3 12:04:07 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 3 May 2010 14:04:07 -0500
Subject: [Freeswitch-users] wrong ptimes and second SDP!,
	cough cough Thomson ST2030
In-Reply-To: <06bafa16896d6e54cc1d5c4e8e5a44bd.squirrel@correo.nodo50.org>
References: <b57c607101b188410fac1458fd0f4edf.squirrel@correo.nodo50.org>
	<31AF5567-8AE7-4ADD-A611-804C6BC6C58E@freeswitch.org>
	<06bafa16896d6e54cc1d5c4e8e5a44bd.squirrel@correo.nodo50.org>
Message-ID: <60E86097-89E6-4D9A-829B-C81FE48C6E35@freeswitch.org>

Might I recommend the Craftsman 16 oz. Rip Claw Hammer?  

/b

On May 3, 2010, at 1:59 PM, Alberto Escudero wrote:

> I set in internal.xml
> 
> <param name="inbound-codec-negotiation" value="scrooge"/>
> 
> But the second SDP returns the ptime to the wrong time.



From peter.olsson at visionutveckling.se  Mon May  3 12:27:48 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 3 May 2010 21:27:48 +0200
Subject: [Freeswitch-users] Strange errors from sqlite DB
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C5576CE4D91@cooper>
References: <549CFEF87AEDE841A38E9D15EAB4C04C5576CE4D91@cooper>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5576CE4D92@cooper>

I've had an instance of FS running on a box for a couple weeks (since I last touched it), it's mostly used for trunking between some lab PBX'es and conferencing, and 2-3 Polycom's also register to it. It was compiled at april 14 from current git at that time (I can look up the exact revision if needed).

Today, after a restart, I suddenly received these errors. I've never seen them before, so I was quite surprised :) Has anyone seen these before? I shut down FS after the error, removed all db-files and restarted again. The error dissapeared, but is this something I should worry about? I've been using FS (very lightly - but still..) for over a year, and this was the first time I noticed it. I found on the list that someone had a similar problem before, but I couldn't find a real explanation to it.

I'm running on Windows 2003.

----

2010-05-03 20:29:21.032501 [ERR] switch_core_sqldb.c:404 SQL ERR [library routine called out of sequence]
insert into sip_authentication (nonce,expires,profile_name,hostname, last_nc) values('6e5df620-d388-4f93-b0b0-8da4cf880ef1', 1272911421, '(NULL)', 'internal', 0)
2010-05-03 20:29:21.048126 [ERR] switch_core_sqldb.c:722 SQL ERR: [select nonce,last_nc from sip_authentication where nonce='6e5df620-d388-4f93-b0b0-8da4cf880ef1' and last_nc < 1] library routine called out of sequence
2010-05-03 20:29:21.048126 [ERR] switch_core_sqldb.c:404 SQL ERR [library routine called out of sequence]
update sip_authentication set expires='1272911421',last_nc=1 where nonce='6e5df620-d388-4f93-b0b0-8da4cf880ef1'
2010-05-03 20:29:21.048126 [ERR] switch_core_sqldb.c:404 SQL ERR [library routine called out of sequence]

Regards,

Peter


From brian at freeswitch.org  Mon May  3 12:33:16 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 3 May 2010 14:33:16 -0500
Subject: [Freeswitch-users] Strange errors from sqlite DB
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C5576CE4D92@cooper>
References: <549CFEF87AEDE841A38E9D15EAB4C04C5576CE4D91@cooper>
	<549CFEF87AEDE841A38E9D15EAB4C04C5576CE4D92@cooper>
Message-ID: <0A4D6EDD-2C4E-4FC1-A48D-187B647ED0BC@freeswitch.org>

Happen to be touching core.db from outside of FreeSWITCH?  Also what SVN/GIT are you on?

/b

On May 3, 2010, at 2:27 PM, Peter Olsson wrote:

> ----
> 
> 2010-05-03 20:29:21.032501 [ERR] switch_core_sqldb.c:404 SQL ERR [library routine called out of sequence]
> insert into sip_authentication (nonce,expires,profile_name,hostname, last_nc) values('6e5df620-d388-4f93-b0b0-8da4cf880ef1', 1272911421, '(NULL)', 'internal', 0)
> 2010-05-03 20:29:21.048126 [ERR] switch_core_sqldb.c:722 SQL ERR: [select nonce,last_nc from sip_authentication where nonce='6e5df620-d388-4f93-b0b0-8da4cf880ef1' and last_nc < 1] library routine called out of sequence
> 2010-05-03 20:29:21.048126 [ERR] switch_core_sqldb.c:404 SQL ERR [library routine called out of sequence]
> update sip_authentication set expires='1272911421',last_nc=1 where nonce='6e5df620-d388-4f93-b0b0-8da4cf880ef1'
> 2010-05-03 20:29:21.048126 [ERR] switch_core_sqldb.c:404 SQL ERR [library routine called out of sequence]
> 
> Regards,
> 
> Peter



From peter.olsson at visionutveckling.se  Mon May  3 12:46:52 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 3 May 2010 21:46:52 +0200
Subject: [Freeswitch-users] Strange errors from sqlite DB
In-Reply-To: <0A4D6EDD-2C4E-4FC1-A48D-187B647ED0BC@freeswitch.org>
References: <549CFEF87AEDE841A38E9D15EAB4C04C5576CE4D91@cooper>
	<549CFEF87AEDE841A38E9D15EAB4C04C5576CE4D92@cooper>,
	<0A4D6EDD-2C4E-4FC1-A48D-187B647ED0BC@freeswitch.org>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5576CE4D93@cooper>

Nope, no scripts, or no other external applications accessing the db's.

I don't have the exact git revision right now, but it was pulled/rebuilt on april 14. I've upgraded to latest git now, but I saved the old binaries and db-files, I will play with it tomorrow and see if it's possible to recreated the error - but as I said, I've never seen this before.

I should also add - the time it hapened FS had just been restarted, one call was running, and two Polycom's trying to register (I guess the sip_authentication table is related to register), so really no load at all.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Brian West [brian at freeswitch.org]
Skickat: den 3 maj 2010 21:33
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Strange errors from sqlite DB

Happen to be touching core.db from outside of FreeSWITCH?  Also what SVN/GIT are you on?

/b

On May 3, 2010, at 2:27 PM, Peter Olsson wrote:

> ----
>
> 2010-05-03 20:29:21.032501 [ERR] switch_core_sqldb.c:404 SQL ERR [library routine called out of sequence]
> insert into sip_authentication (nonce,expires,profile_name,hostname, last_nc) values('6e5df620-d388-4f93-b0b0-8da4cf880ef1', 1272911421, '(NULL)', 'internal', 0)
> 2010-05-03 20:29:21.048126 [ERR] switch_core_sqldb.c:722 SQL ERR: [select nonce,last_nc from sip_authentication where nonce='6e5df620-d388-4f93-b0b0-8da4cf880ef1' and last_nc < 1] library routine called out of sequence
> 2010-05-03 20:29:21.048126 [ERR] switch_core_sqldb.c:404 SQL ERR [library routine called out of sequence]
> update sip_authentication set expires='1272911421',last_nc=1 where nonce='6e5df620-d388-4f93-b0b0-8da4cf880ef1'
> 2010-05-03 20:29:21.048126 [ERR] switch_core_sqldb.c:404 SQL ERR [library routine called out of sequence]
>
> Regards,
>
> Peter


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

!DSPAM:4bdf268d32931789114366!



From pjintheusa at gmail.com  Mon May  3 12:52:39 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Mon, 3 May 2010 15:52:39 -0400
Subject: [Freeswitch-users] mobile phone clients / fs cluster
In-Reply-To: <4BDB6FA4.10606@todandlorna.com>
References: <s2p367751821004291356lc7a5c591m556b1f48ebc3a052@mail.gmail.com>
	<p2sb74751491004292211i4d66c1d2me49ffcfc18852692@mail.gmail.com>
	<y2m367751821004301526r17a57e70o45ec9ba275539095@mail.gmail.com>
	<4BDB6FA4.10606@todandlorna.com>
Message-ID: <o2n367751821005031252h53dad5b4n92632250e6ee11de@mail.gmail.com>

Tod,

Thanks for your reply:

>>> The 3G networks are just like connecting wirelessly to the internet.
Depending on your provider, your phone gets an internet addressable address,
or just an internal network address which is NATed before going to the
internet

It appears to me that the if a mobile device contacts FS1,  the mobile
device accepts ip packets from FS1. But NOT from FS2. And vice versa.

The invite from FS1 and FS2 are exactly the same (except for the IP address
of course).

So let me ask, can a mobile device receive unsolicited packets from the
Internet? Or are these restricted? If they are restricted, the above makes
sense.


Pj



On Fri, Apr 30, 2010 at 8:02 PM, Tod Hansmann <freeswitch.org@
todandlorna.com> wrote:

>  Phil (Or do you prefer Phillip?),
>
> You will need to probably draw this out a bit.  What is the path by which
> the home office phones connect to the FS boxes?  What path do the cell
> phones take?  What routes do you have for the return data on each box?  Are
> any NATs/Firewalls involved?
>
> I think this is firmly a networking question.  The proxy might come into
> play as well here.  The 3G networks are just like connecting wirelessly to
> the internet.  Depending on your provider, your phone gets an internet
> addressable address, or just an internal network address which is NATed
> before going to the internet.  That NAT can be tested, if you have the right
> tools on your phone and what you're connecting to, but that will change from
> provider to provider and maybe even day to day, location to location.
>
> That should be enough to start thinking about the problem and where it
> might lay.
>
> Cheers,
>
> Tod Hansmann
>
>
> On 4/30/2010 4:26 PM, Phillip Jones wrote:
>
> Ram,
>
>  I think thats what I am asking? I am not sure how the 3G network works.
> Whether there are restrictions on how servers can communicate to clients
> etc.
>
>  Perhaps I am just way of base also. I don't know. TGIF.
>
>  Pj
>
>
> On Fri, Apr 30, 2010 at 1:11 AM, ram <talk2ram at gmail.com> wrote:
>
>> Hi
>>
>> why not its possible
>>
>>  Ram
>>
>>   On Fri, Apr 30, 2010 at 2:26 AM, Phillip Jones <pjintheusa at gmail.com>wrote:
>>
>>>  Hi there,
>>>
>>> I have two FS servers (FS1 and FS2) behind an inbound OpenSIPS proxy.
>>> Outbound (terminating) traffic goes directly from each FS box.
>>>
>>> All my home office phones get calls no matter which box they are
>>> registered with or which box the call comes in on - thanks Anthony.
>>>
>>> However my SIP client on various iPhone/Androids etc only receive calls
>>> that originate on the box on which they are registered.
>>>
>>> Looking at the SIP trace - when the call comes in on the 'wrong' box, the
>>> invites to these SIP clients do not even get a response. Presumably because
>>> the 3G network has no idea who this new IP in the "from address" is, who
>>> trying to contact them.
>>>
>>> Question is, is there a way around this - our will I have start routing
>>> all terminating traffic out through the proxy also.
>>>
>>> Any insight appreciated.
>>>
>>> Thanks!
>>>
>>> Pj
>>>
>>>
>>>  _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From peter.olsson at visionutveckling.se  Mon May  3 12:52:14 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 3 May 2010 21:52:14 +0200
Subject: [Freeswitch-users] Strange errors from sqlite DB
In-Reply-To: <0A4D6EDD-2C4E-4FC1-A48D-187B647ED0BC@freeswitch.org>
References: <549CFEF87AEDE841A38E9D15EAB4C04C5576CE4D91@cooper>
	<549CFEF87AEDE841A38E9D15EAB4C04C5576CE4D92@cooper>,
	<0A4D6EDD-2C4E-4FC1-A48D-187B647ED0BC@freeswitch.org>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5576CE4D95@cooper>

Got the git revision - here it is: git-d6ee682 2010-04-13 13:38:47 -0700

Thanks,

Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Brian West [brian at freeswitch.org]
Skickat: den 3 maj 2010 21:33
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Strange errors from sqlite DB

Happen to be touching core.db from outside of FreeSWITCH?  Also what SVN/GIT are you on?

/b

On May 3, 2010, at 2:27 PM, Peter Olsson wrote:

> ----
>
> 2010-05-03 20:29:21.032501 [ERR] switch_core_sqldb.c:404 SQL ERR [library routine called out of sequence]
> insert into sip_authentication (nonce,expires,profile_name,hostname, last_nc) values('6e5df620-d388-4f93-b0b0-8da4cf880ef1', 1272911421, '(NULL)', 'internal', 0)
> 2010-05-03 20:29:21.048126 [ERR] switch_core_sqldb.c:722 SQL ERR: [select nonce,last_nc from sip_authentication where nonce='6e5df620-d388-4f93-b0b0-8da4cf880ef1' and last_nc < 1] library routine called out of sequence
> 2010-05-03 20:29:21.048126 [ERR] switch_core_sqldb.c:404 SQL ERR [library routine called out of sequence]
> update sip_authentication set expires='1272911421',last_nc=1 where nonce='6e5df620-d388-4f93-b0b0-8da4cf880ef1'
> 2010-05-03 20:29:21.048126 [ERR] switch_core_sqldb.c:404 SQL ERR [library routine called out of sequence]
>
> Regards,
>
> Peter


_______________________________________________
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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From anthony.minessale at gmail.com  Mon May  3 13:00:32 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 3 May 2010 15:00:32 -0500
Subject: [Freeswitch-users] Trouble With Transfer and Async Event Socket
In-Reply-To: <012d01caeaed$47fc3460$d7f49d20$@com>
References: <n2sf1b753861005030733u73a51a2fh4fae23812f5089@mail.gmail.com>
	<n2y191c3a031005031105v4cd7ad3by96e59cbfc19b319@mail.gmail.com>
	<012d01caeaed$47fc3460$d7f49d20$@com>
Message-ID: <AANLkTikZQllnCkEourWKLCu1iykRBBNzKuQazoBt8SVc@mail.gmail.com>

well that's because it's what "both" implies, the "current" leg and the one
it's bridged to.
if you are not bridged anymore, naturally, it won't work.



On Mon, May 3, 2010 at 1:20 PM, Joseph Crivello <josephcrivello at gmail.com>wrote:

>  Example command:
>
>
>
> Command: transfer
>
> Arguments: -both ConfXfer-1 XML default
>
>
>
> And here is the referenced dialplan extension:
>
>
>
>     <extension name="ConfXfer">
>
>       <condition field="destination_number" expression="^ConfXfer-(\d+)$">
>
>         <action application="answer"/>
>
>         <action application="conference" data="$1" />
>
>       </condition>
>
>     </extension>
>
>
>
> Interestingly, I recently discovered that the transfer works if I do it
> before the bridge finishes (figured that out by accident).
>
>
>
> -Joe
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Monday, May 03, 2010 1:06 PM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Trouble With Transfer and Async Event
> Socket
>
>
>
> What command are you using to try to transfer it to your conference?
>
>
>
> On Mon, May 3, 2010 at 9:33 AM, Joe Crivello <josephcrivello at gmail.com>
> wrote:
>
> Hello All,
>
> I am currently working on a conference bridge based on FreeSWITCH that
> allows some conference members to dial out to a prompted number.
>
> Currently my implementation binds * in conference.xml to the "transfer"
> action, leading to a dial plan extension that runs an async full outgoing
> event socket. My application serving the event socket does the following:
>
> 1) Verifies the user is allowed to dial out, and writes a record to the
> database.
> 2) Retrieves the number to be dialed with 'play_and_get_digits'
> 3) Bridges the call to the dialed number
> 4) Waits for the a-leg to press star
> 5) Uses 'transfer' to transfer both legs back to the conference
>
> Everything works up to step #5, which does nothing.
>
> There is no NOTICE outputted to the console indicating that a transfer is
> taking place and neither leg is rejoined to the conference.
>
> I get a normal event back after the transfer sendmsg:
>
> Event-Name: SOCKET_DATA
> Content-Type: command/reply
> Reply-Text: %2BOK
>
> If I subscribe to all events before running the transfer and receive events
> in a loop, I receive nothing but routine RE_SCHEDULE and CALL_UPDATE events
> after the transfer.
>
> I am using ESL to communicate with FreeSWITCH, and prior to running any of
> steps #1-5 I set the event lock to true. The line from the dial plan
> extension that calls my application follows:
>
> <action application="socket" data="172.18.11.1:8001 async full" />
>
> Any ideas? Perhaps I'm "doing it wrong"? I am very new to FS.
>
> Thanks!
> Joe Crivello
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <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
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From peter.olsson at visionutveckling.se  Mon May  3 12:58:03 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 3 May 2010 21:58:03 +0200
Subject: [Freeswitch-users] Strange errors from sqlite DB
In-Reply-To: <0A4D6EDD-2C4E-4FC1-A48D-187B647ED0BC@freeswitch.org>
References: <549CFEF87AEDE841A38E9D15EAB4C04C5576CE4D91@cooper>
	<549CFEF87AEDE841A38E9D15EAB4C04C5576CE4D92@cooper>,
	<0A4D6EDD-2C4E-4FC1-A48D-187B647ED0BC@freeswitch.org>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5576CE4D96@cooper>

Brian,

This is the log when it first occured. I'm not sure if I'll ever recreate the issue, so do you think I should open a jira anyway? This log is more or less all I have.

2010-05-03 20:25:32.673126 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/50711 at 10.10.1.35) State Change CS_ROUTING -> CS_EXECUTE
2010-05-03 20:25:32.673126 [DEBUG] switch_core_session.c:1021 Send signal sofia/internal/50711 at 10.10.1.35 [BREAK]
2010-05-03 20:25:32.673126 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/50711 at 10.10.1.35) State ROUTING going to sleep
2010-05-03 20:25:32.673126 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/50711 at 10.10.1.35) Running State Change CS_EXECUTE
2010-05-03 20:25:32.673126 [DEBUG] switch_core_state_machine.c:348 (sofia/internal/50711 at 10.10.1.35) State EXECUTE
2010-05-03 20:25:32.673126 [DEBUG] mod_sofia.c:226 sofia/internal/50711 at 10.10.1.35 SOFIA EXECUTE
2010-05-03 20:25:32.673126 [DEBUG] switch_core_state_machine.c:157 sofia/internal/50711 at 10.10.1.35 Standard EXECUTE
EXECUTE sofia/internal/50711 at 10.10.1.35 set(hangup_after_bridge=true)
2010-05-03 20:25:32.673126 [DEBUG] mod_dptools.c:816 sofia/internal/50711 at 10.10.1.35 SET [hangup_after_bridge]=[true]
EXECUTE sofia/internal/50711 at 10.10.1.35 bridge(sofia/gateway/vip2000-trunk/00709275071)
2010-05-03 20:25:32.673126 [NOTICE] switch_channel.c:669 New Channel sofia/internal/00709275071 [41b995f9-df54-4053-a521-ae804c6c1c6b]
2010-05-03 20:25:32.673126 [DEBUG] mod_sofia.c:3386 (sofia/internal/00709275071) State Change CS_NEW -> CS_INIT
2010-05-03 20:25:32.673126 [DEBUG] switch_core_session.c:1021 Send signal sofia/internal/00709275071 [BREAK]
2010-05-03 20:25:32.673126 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/00709275071) Running State Change CS_INIT
2010-05-03 20:25:32.673126 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/00709275071) State INIT
2010-05-03 20:25:32.673126 [DEBUG] mod_sofia.c:83 sofia/internal/00709275071 SOFIA INIT
2010-05-03 20:25:32.673126 [DEBUG] mod_sofia.c:117 (sofia/internal/00709275071) State Change CS_INIT -> CS_ROUTING
2010-05-03 20:25:32.673126 [DEBUG] switch_core_session.c:1021 Send signal sofia/internal/00709275071 [BREAK]
2010-05-03 20:25:32.673126 [DEBUG] switch_core_state_machine.c:338 (sofia/internal/00709275071) State INIT going to sleep
2010-05-03 20:25:32.673126 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/00709275071) Running State Change CS_ROUTING
2010-05-03 20:25:32.673126 [DEBUG] sofia.c:4153 Channel sofia/internal/00709275071 entering state [calling][0]
2010-05-03 20:25:32.673126 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/00709275071) State ROUTING
2010-05-03 20:25:32.673126 [DEBUG] mod_sofia.c:140 sofia/internal/00709275071 SOFIA ROUTING
2010-05-03 20:25:32.673126 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/00709275071) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2010-05-03 20:25:32.673126 [DEBUG] switch_core_session.c:1021 Send signal sofia/internal/00709275071 [BREAK]
2010-05-03 20:25:32.673126 [DEBUG] switch_core_state_machine.c:341 (sofia/internal/00709275071) State ROUTING going to sleep
2010-05-03 20:25:32.673126 [DEBUG] switch_core_state_machine.c:314 (sofia/internal/00709275071) Running State Change CS_CONSUME_MEDIA
2010-05-03 20:25:32.673126 [DEBUG] switch_core_state_machine.c:360 (sofia/internal/00709275071) State CONSUME_MEDIA
2010-05-03 20:25:32.673126 [DEBUG] switch_core_state_machine.c:360 (sofia/internal/00709275071) State CONSUME_MEDIA going to sleep
2010-05-03 20:25:32.688751 [INFO] sofia.c:662 Update Callee ID to "00709275071" <00709275071>
2010-05-03 20:25:32.751251 [ERR] switch_core_sqldb.c:722 SQL ERR: [select sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'CS_ROUTING','unknown','10.10.1.35',sip_presence.status,sip_presence.rpid from sip_subscriptions left join sip_presence on (sip_subscriptions.sub_to_user=sip_presence.sip_user and sip_subscriptions.sub_to_host=sip_presence.sip_host and sip_subscriptions.profile_name=sip_presence.profile_name) where sip_subscriptions.expires > -1 and (event='presence' or event='dialog') and sub_to_user='50711' and (sub_to_host='10.10.1.35' or presence_hosts like '%10.10.1.35%') and (sip_subscriptions.profile_name = 'internal' or sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host)] library routine called out of sequence
2010-05-03 20:25:32.782501 [DEBUG] sofia.c:4153 Channel sofia/internal/00709275071 entering state [proceeding][180]
2010-05-03 20:25:32.782501 [NOTICE] sofia.c:4223 Ring-Ready sofia/internal/00709275071!
2010-05-03 20:25:32.782501 [NOTICE] mod_sofia.c:1837 Ring-Ready sofia/internal/50711 at 10.10.1.35!
2010-05-03 20:25:32.782501 [ERR] switch_core_sqldb.c:525 Statement Error [select 'appearance-index=1' from sip_subscriptions where expires > -1 and hostname='teller' and event='call-info' and sub_to_user='00709275071' and sub_to_host='192.168.1.55']!
2010-05-03 20:25:32.782501 [DEBUG] switch_core_session.c:641 Send signal sofia/internal/50711 at 10.10.1.35 [BREAK]
2010-05-03 20:25:32.782501 [NOTICE] switch_ivr_originate.c:437 Ring Ready sofia/internal/50711 at 10.10.1.35!
2010-05-03 20:25:32.782501 [DEBUG] sofia.c:4153 Channel sofia/internal/00709275071 entering state [proceeding][183]
2010-05-03 20:25:32.782501 [DEBUG] sofia.c:4164 Remote SDP:

________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Brian West [brian at freeswitch.org]
Skickat: den 3 maj 2010 21:33
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Strange errors from sqlite DB

Happen to be touching core.db from outside of FreeSWITCH?  Also what SVN/GIT are you on?

/b

On May 3, 2010, at 2:27 PM, Peter Olsson wrote:

> ----
>
> 2010-05-03 20:29:21.032501 [ERR] switch_core_sqldb.c:404 SQL ERR [library routine called out of sequence]
> insert into sip_authentication (nonce,expires,profile_name,hostname, last_nc) values('6e5df620-d388-4f93-b0b0-8da4cf880ef1', 1272911421, '(NULL)', 'internal', 0)
> 2010-05-03 20:29:21.048126 [ERR] switch_core_sqldb.c:722 SQL ERR: [select nonce,last_nc from sip_authentication where nonce='6e5df620-d388-4f93-b0b0-8da4cf880ef1' and last_nc < 1] library routine called out of sequence
> 2010-05-03 20:29:21.048126 [ERR] switch_core_sqldb.c:404 SQL ERR [library routine called out of sequence]
> update sip_authentication set expires='1272911421',last_nc=1 where nonce='6e5df620-d388-4f93-b0b0-8da4cf880ef1'
> 2010-05-03 20:29:21.048126 [ERR] switch_core_sqldb.c:404 SQL ERR [library routine called out of sequence]
>
> Regards,
>
> Peter


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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!DSPAM:4bdf268d32931789114366!



From mike at jerris.com  Mon May  3 13:14:24 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 3 May 2010 16:14:24 -0400
Subject: [Freeswitch-users] mod_xml_cdr error log is not working
In-Reply-To: <l2z7abab2411005030343xdbe1827coc7e531214aa78864@mail.gmail.com>
References: <g2p7abab2411004291856s2c2d5018g103f33cf2be9dd4b@mail.gmail.com>
	<DD0949EC-0C93-4A8F-9EA4-C6A669AE93FF@jerris.com>
	<l2z7abab2411005030343xdbe1827coc7e531214aa78864@mail.gmail.com>
Message-ID: <68DF5166-42A6-4316-9F96-3FD3CC6C9B62@jerris.com>

would you mind making sure this gets documented on the wiki, and, if it is not in the sample configuration, send us a patch for that via http://jira.freeswitch.org ?

Mike

On May 3, 2010, at 6:43 AM, Frank Church wrote:

> I have received some help about it in http://lists.freeswitch.org/pipermail/freeswitch-users/2010-May/057453.html.
> 
> It requires an additional undocumented parameter in mod_xml_cdr.conf, though that may not have been necessary in earlier versions.
> 
> 
> On 3 May 2010 06:06, Michael Jerris <mike at jerris.com> wrote:
> the defaults in windows are all relative to the running freeswitch base dir, unless you explicitly set them.  there may be some issue with trailing vs no trailing path seperator when you configure them.
> 
> Mike
> 
> On Apr 29, 2010, at 9:56 PM, Frank Church wrote:
> 
> > I have set the err-log-dir and log-dr in xml_cdr.conf.xml but the logs
> > are not working. The system is a windows system and I wonder if the
> > defaults for windows are different.
> >
> > I have logs/xml_cdr  in addition to the  log/xml_cdr in the
> > c:\freeswitch directory but Freeswitch can't find them.
> >
> > Logs snippet
> > =========
> > 2010-04-30 02:41:02.984375 [DEBUG] switch_core_state_machine.c:314
> > (sofia/internal/1811 at 192.168.1.133) Running State Change CS_REPORTING
> > 2010-04-30 02:41:02.984375 [DEBUG] switch_core_state_machine.c:590
> > (sofia/internal/1811 at 192.168.1.133) State REPORTING
> > 2010-04-30 02:41:03.812500 [ERR] mod_xml_cdr.c:359 Got error [0]
> > posting to web server [http://192.168.1.20:8132/]
> > 2010-04-30 02:41:03.812500 [ERR] mod_xml_cdr.c:366 Retry will be with
> > url [http://192.168.1.20:8132/]
> > 2010-04-30 02:41:03.812500 [ERR] mod_xml_cdr.c:377 Unable to post to
> > web server, writing to file
> > 2010-04-30 02:41:03.812500 [ERR] mod_xml_cdr.c:399 Error![No such file
> > or directory]
> >

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From mike at jerris.com  Mon May  3 13:15:36 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 3 May 2010 16:15:36 -0400
Subject: [Freeswitch-users] Need help on realtime integration Freeswitch
	with Opensips.
In-Reply-To: <s2jd9f1e4661005030020qb61f7128o57fe42aaa2cfb3c3@mail.gmail.com>
References: <s2jd9f1e4661005030020qb61f7128o57fe42aaa2cfb3c3@mail.gmail.com>
Message-ID: <8E1C56DF-7D1C-4E23-B3F8-1838851EBD19@jerris.com>

Take a look at http://wiki.freeswitch.org/wiki/Xml_curl .  This in combination with odbc should address all realtime concerns.

Mike

On May 3, 2010, at 3:20 AM, hung nguyen wrote:

> Hi list.
> I have deployed cluster FS with share DB - using ODBC in core.
> I think it is great if we can deploy  realtime integration Freeswitch with Opensips. With this, Opensips will act as register,proxy server and load balancer in front of FS boxs.
> Anybody have same ideal and done this.
> Tks.
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From mike at jerris.com  Mon May  3 13:17:12 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 3 May 2010 16:17:12 -0400
Subject: [Freeswitch-users] Trouble With 'transfer' DP After a 'bridge'
	on Async Event Socket
In-Reply-To: <07a101caeab4$dd343700$979ca500$@com>
References: <07a101caeab4$dd343700$979ca500$@com>
Message-ID: <051C376A-BFB0-4A90-9804-2684EEA006B2@jerris.com>

try uuid_transfer api ?

Mike

On May 3, 2010, at 7:36 AM, Joseph Crivello wrote:

> Hello All,
> 
> I am currently working on a conference bridge based on FreeSWITCH that
> allows some conference members to dial out to a prompted number.
> 
> Currently my implementation binds * in conference.xml to the "transfer"
> action, leading to a dial plan extension that runs an async full outgoing
> event socket. My application serving the event socket does the following:
> 
> 1) Verifies the user is allowed to dial out, and writes a record to the
> database.
> 2) Retrieves the number to be dialed with 'play_and_get_digits'
> 3) Bridges the call to the dialed number
> 4) Waits for the a-leg to press star
> 5) Uses 'transfer' to transfer both legs back to the conference 
> 
> Everything works up to step #5, which does nothing. 
> 
> There is no NOTICE outputted to the console indicating that a transfer is
> taking place and neither leg is rejoined to the conference.
> 
> I get a normal event back after the transfer sendmsg:
> 
> Event-Name: SOCKET_DATA
> Content-Type: command/reply
> Reply-Text: %2BOK
> 
> If I subscribe to all events before running the transfer and receive events
> in a loop, I receive nothing but routine RE_SCHEDULE and CALL_UPDATE events
> after the transfer.
> 
> I am using ESL to communicate with FreeSWITCH, and prior to running any of
> steps #1-5 I set the event lock to true. The line from the dial plan
> extension that calls my application follows:
> 
> <action application="socket" data="172.18.11.1:8001 async full" />
> 
> Any ideas? Perhaps I'm "doing it wrong"? I am very new to FS.



From mike at jerris.com  Mon May  3 13:20:47 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 3 May 2010 16:20:47 -0400
Subject: [Freeswitch-users] External SIP profile
In-Reply-To: <56DF79F1-2709-49CC-B5A3-C3A733F2B26E@binarysignal.com>
References: <56DF79F1-2709-49CC-B5A3-C3A733F2B26E@binarysignal.com>
Message-ID: <BE1DE3B7-8C40-41E3-B8FE-169A4E9423C3@jerris.com>

yes, roughly the same as a softphone, I think the mitel had some issues with some of our packets, try callerid_in_from 

<!--/// replace the INVITE from user with the channel's caller-id ///-->
<!--/// this is false by default because outbound calls generally need username in the INVITE ///-->
<param name="caller-id-in-from" value="false"/>

or experiment with other settings from the wiki:

http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files

more info on mitel devices:

http://wiki.freeswitch.org/wiki/Interop_List#Mitel_devices

Mike


On May 3, 2010, at 12:30 PM, Kurt Ward wrote:

> Maybe this is a bit of an uneducated noob question (or  
> misunderstanding about SIP) but here goes:
> 
> If I have an external profile set up and registering, does it  
> essentially behave like a soft phone would behave?
> 
> What I am running into is incoming calls work fine, but outgoing calls  
> always produce a 404.  If I use a soft phone using the same  
> registration I can make calls with no problem (in both scenarios the  
> calls are to/from 4 digit extensions on a Mitel PBX). 
> 

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From josephcrivello at gmail.com  Mon May  3 13:22:12 2010
From: josephcrivello at gmail.com (Joseph Crivello)
Date: Mon, 3 May 2010 15:22:12 -0500
Subject: [Freeswitch-users] Trouble With Transfer and Async Event Socket
In-Reply-To: <AANLkTikZQllnCkEourWKLCu1iykRBBNzKuQazoBt8SVc@mail.gmail.com>
References: <n2sf1b753861005030733u73a51a2fh4fae23812f5089@mail.gmail.com>	<n2y191c3a031005031105v4cd7ad3by96e59cbfc19b319@mail.gmail.com>	<012d01caeaed$47fc3460$d7f49d20$@com>
	<AANLkTikZQllnCkEourWKLCu1iykRBBNzKuQazoBt8SVc@mail.gmail.com>
Message-ID: <01ae01caeafe$5168b5d0$f43a2170$@com>

I think I confused the situation with my ending comment in my last email. 

 

The call is bridged and in the CS_EXCHANGE_MEDIA state when I run the
transfer.

 

What I meant to say in my ending comment in my last email was that I noticed
if I run the transfer command as listed below when the B-leg is not yet in
the CS_EXCHANGE_MEDIA state, the transfer works (although of course a
WARNING is outputted to the console complaining that there is no B-leg).

 

-Joe

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, May 03, 2010 3:01 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Trouble With Transfer and Async Event Socket

 

well that's because it's what "both" implies, the "current" leg and the one
it's bridged to.

if you are not bridged anymore, naturally, it won't work.

 

 

On Mon, May 3, 2010 at 1:20 PM, Joseph Crivello <josephcrivello at gmail.com>
wrote:

Example command:

 

Command: transfer

Arguments: -both ConfXfer-1 XML default

 

And here is the referenced dialplan extension:

 

    <extension name="ConfXfer">

      <condition field="destination_number" expression="^ConfXfer-(\d+)$">

        <action application="answer"/>

        <action application="conference" data="$1" />

      </condition>

    </extension>

 

Interestingly, I recently discovered that the transfer works if I do it
before the bridge finishes (figured that out by accident).

 

-Joe

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, May 03, 2010 1:06 PM


To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Trouble With Transfer and Async Event Socket

 

What command are you using to try to transfer it to your conference?

 

On Mon, May 3, 2010 at 9:33 AM, Joe Crivello <josephcrivello at gmail.com>
wrote:

Hello All,

I am currently working on a conference bridge based on FreeSWITCH that
allows some conference members to dial out to a prompted number.

Currently my implementation binds * in conference.xml to the "transfer"
action, leading to a dial plan extension that runs an async full outgoing
event socket. My application serving the event socket does the following:

1) Verifies the user is allowed to dial out, and writes a record to the
database.
2) Retrieves the number to be dialed with 'play_and_get_digits'
3) Bridges the call to the dialed number
4) Waits for the a-leg to press star
5) Uses 'transfer' to transfer both legs back to the conference

Everything works up to step #5, which does nothing.

There is no NOTICE outputted to the console indicating that a transfer is
taking place and neither leg is rejoined to the conference.

I get a normal event back after the transfer sendmsg:

Event-Name: SOCKET_DATA
Content-Type: command/reply
Reply-Text: %2BOK

If I subscribe to all events before running the transfer and receive events
in a loop, I receive nothing but routine RE_SCHEDULE and CALL_UPDATE events
after the transfer.

I am using ESL to communicate with FreeSWITCH, and prior to running any of
steps #1-5 I set the event lock to true. The line from the dial plan
extension that calls my application follows:

<action application="socket" data=" <http://172.18.11.1:8001/>
172.18.11.1:8001 async full" />

Any ideas? Perhaps I'm "doing it wrong"? I am very new to FS.

Thanks!
Joe Crivello 
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
Anthony Minessale II

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

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

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


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




-- 
Anthony Minessale II

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

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

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

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From peter.olsson at visionutveckling.se  Mon May  3 13:36:18 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 3 May 2010 22:36:18 +0200
Subject: [Freeswitch-users] run error after building in vs 2008!
In-Reply-To: <E141A983-9EC5-4C62-8D17-675DC0A67009@jerris.com>
References: <i2v1fdf92721004302253l7dbdf254p1ff2b3884ec5a4d4@mail.gmail.com>
	<l2o367751821005010612j46281f6fg17629c38bcd07f4@mail.gmail.com>
	<n2i1fdf92721005011131o221b6dd4sf483e00613963d5@mail.gmail.com>
	<x2z367751821005011229g3b6c217kbda766daac32a44d@mail.gmail.com>,
	<E141A983-9EC5-4C62-8D17-675DC0A67009@jerris.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5576CE4D97@cooper>

I also came across this error today on a Windows 2003 server (same issue as XP). For now I've commented out the call to inet_ntop(), but I think the code below probably could be used - I Googled it. I'll try to find some time to get it working within FS. I think there are still some XP/2003 boxes out there, so it should be worth fixing.

/Peter

/* const char *
 * inet_ntop6(src, dst, size)
 *	convert IPv6 binary address into presentation (printable) format
 * author:
 *	Paul Vixie, 1996.
 */

static const char *
inet_ntop6(src, dst, size)
	const u_char *src;
	char *dst;
	size_t size;
{
	/*
	 * Note that int32_t and int16_t need only be "at least" large enough
	 * to contain a value of the specified size.  On some systems, like
	 * Crays, there is no such thing as an integer variable with 16 bits.
	 * Keep this in mind if you think this function should have been coded
	 * to use pointer overlays.  All the world's not a VAX.
	 */
	char tmp[sizeof "ffff:ffff:ffff:ffff:ffff:ffff:255.255.255.255"], *tp;
	struct { int base, len; } best, cur;
	u_int words[IN6ADDRSZ / INT16SZ];
	int i;

	/*
	 * Preprocess:
	 *	Copy the input (bytewise) array into a wordwise array.
	 *	Find the longest run of 0x00's in src[] for :: shorthanding.
	 */
	memset(words, '\0', sizeof words);
	for (i = 0; i < IN6ADDRSZ; i++)
		words[i / 2] |= (src[i] << ((1 - (i % 2)) << 3));
	best.base = -1;
	cur.base = -1;
	for (i = 0; i < (IN6ADDRSZ / INT16SZ); i++) {
		if (words[i] == 0) {
			if (cur.base == -1)
				cur.base = i, cur.len = 1;
			else
				cur.len++;
		} else {
			if (cur.base != -1) {
				if (best.base == -1 || cur.len > best.len)
					best = cur;
				cur.base = -1;
			}
		}
	}
	if (cur.base != -1) {
		if (best.base == -1 || cur.len > best.len)
			best = cur;
	}
	if (best.base != -1 && best.len < 2)
		best.base = -1;

	/*
	 * Format the result.
	 */
	tp = tmp;
	for (i = 0; i < (IN6ADDRSZ / INT16SZ); i++) {
		/* Are we inside the best run of 0x00's? */
		if (best.base != -1 && i >= best.base &&
		    i < (best.base + best.len)) {
			if (i == best.base)
				*tp++ = ':';
			continue;
		}
		/* Are we following an initial run of 0x00s or any real hex? */
		if (i != 0)
			*tp++ = ':';
		/* Is this address an encapsulated IPv4? */
		if (i == 6 && best.base == 0 &&
		    (best.len == 6 || (best.len == 5 && words[5] == 0xffff))) {
			if (!inet_ntop4(src+12, tp, sizeof tmp - (tp - tmp)))
				return (NULL);
			tp += strlen(tp);
			break;
		}
		tp += SPRINTF((tp, "%x", words[i]));
	}
	/* Was it a trailing run of 0x00's? */
	if (best.base != -1 && (best.base + best.len) == (IN6ADDRSZ / INT16SZ))
		*tp++ = ':';
	*tp++ = '\0';

	/*
	 * Check for overflow, copy, and we're done.
	 */
	if ((size_t)(tp - tmp) > size) {
		errno = ENOSPC;
		return (NULL);
	}
	strcpy(dst, tmp);
	return (dst);
}

________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Michael Jerris [mike at jerris.com]
Skickat: den 3 maj 2010 06:04
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] run error after building in vs 2008!

This was just added in :

commit f1430d521a767a13035d6d8e96686564552194fd
Author: Anthony Minessale <anthm at freeswitch.org<mailto:anthm at freeswitch.org>>
Date:   Fri Apr 30 15:01:32 2010 -0500

    fix switch_get_addr to work with v6 properly

to fix ipv6 support.  We welcome a patch to restore xp support.

Mike

On May 1, 2010, at 3:29 PM, Phillip Jones wrote:

Oh yeah - looks like that call is not supported in older os like XP.

http://social.msdn.microsoft.com/Forums/en-US/vcgeneral/thread/e40465f2-41b7-4243-ad33-15ae9366f4e6

FreeSWITCH does support XP though - http://wiki.freeswitch.org/wiki/Users_Guide_Introduction

So I don't know.

On Sat, May 1, 2010 at 2:31 PM, babak yakhchali <babak.freeswitch at gmail.com<mailto:babak.freeswitch at gmail.com>> wrote:
ya I've got that in place
but as I searched the web they say it's related to windows version (xp vista . . .)

!DSPAM:4bde4cca32933312412468!


From peter.olsson at visionutveckling.se  Mon May  3 13:43:56 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 3 May 2010 22:43:56 +0200
Subject: [Freeswitch-users] Suggestions for creating diff's in git...
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5576CE4D98@cooper>

I've been using SVN for a couple of years, but I'm quite new to git (as new as the FS project).

I haven't yet found a good (easy) way to create diff's, to send up to jira cases, does anyone have a good tutorial for this?

I've made some patches, but I haven't submitted them yet because of this.

Regards,

Peter


From anthony.minessale at gmail.com  Mon May  3 13:48:30 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 3 May 2010 15:48:30 -0500
Subject: [Freeswitch-users] Anyone have analog Fax working through FS
	and Sangoma cards?
In-Reply-To: <83B388ED98C64D23A32EBC8828A68E19@greyhawk.tonecommander.com>
References: <Acrq8cSCURuBYfCsRGe9uS/am0Xr2A==>
	<83B388ED98C64D23A32EBC8828A68E19@greyhawk.tonecommander.com>
Message-ID: <AANLkTinka6gTDxwVHI9wuaU6wYJo6woWCd8CWgmf1yjd@mail.gmail.com>

Do you mean passing faxes through?
you probably need

<param name="rtp-autoflush-during-bridge" value="false"/>

and

<param name="rtp-timer-name" value="none"/>

in your profile


On Mon, May 3, 2010 at 1:52 PM, Robert Hadley <robert.hadley at teotech.com>wrote:

>
>
> Does anyone have analog Fax working through Freeswitch and Sangoma A101 &
> A200/FXS cards?  If so, would you share what configuration and dialplan
> settings need to be made?  I have a problem receiving Faxes, which only work
> 80% of the time and often take 10-12 minutes per Fax, versus 2 minutes when
> using the Fax machine on analog POTS line.
>
>
>
> My setup: SendingFax --- {PSTN} --- PRI --- A101 --- FS --- A200/FXS ---
> MyFax
>
>
>
>
>
> Pastebin to dialplan, wanpipe conf, vars.xml, internal.xml:
> http://pastebin.freeswitch.org/12884
>
>
>
>
>
> Any help would be greatly appreciated.
>
>
>
> Thanks,
>
> Robert
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <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
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From brian at freeswitch.org  Mon May  3 13:50:48 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 3 May 2010 15:50:48 -0500
Subject: [Freeswitch-users] Suggestions for creating diff's in git...
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C5576CE4D98@cooper>
References: <549CFEF87AEDE841A38E9D15EAB4C04C5576CE4D98@cooper>
Message-ID: <8E40D6E4-05BE-49D4-8BED-BECA792D6B7A@freeswitch.org>

git diff usually works well.

/b

On May 3, 2010, at 3:43 PM, Peter Olsson wrote:

> I've been using SVN for a couple of years, but I'm quite new to git (as new as the FS project).
> 
> I haven't yet found a good (easy) way to create diff's, to send up to jira cases, does anyone have a good tutorial for this?
> 
> I've made some patches, but I haven't submitted them yet because of this.
> 
> Regards,
> 
> Peter



From anthony.minessale at gmail.com  Mon May  3 13:51:42 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 3 May 2010 15:51:42 -0500
Subject: [Freeswitch-users] Trouble With Transfer and Async Event Socket
In-Reply-To: <01ae01caeafe$5168b5d0$f43a2170$@com>
References: <n2sf1b753861005030733u73a51a2fh4fae23812f5089@mail.gmail.com>
	<n2y191c3a031005031105v4cd7ad3by96e59cbfc19b319@mail.gmail.com>
	<012d01caeaed$47fc3460$d7f49d20$@com>
	<AANLkTikZQllnCkEourWKLCu1iykRBBNzKuQazoBt8SVc@mail.gmail.com>
	<01ae01caeafe$5168b5d0$f43a2170$@com>
Message-ID: <AANLkTimW88uzO9GDBilI8VEvas1D8m3wFCtpgLJN_H2G@mail.gmail.com>

Your attempt to underscore the area where you are having the problem has
obscured the necessary details to help you.

You need to re-describe with more details.

you might want to use the api interface on your socket app and use
uuid_transfer -both <uuid of a leg>

On Mon, May 3, 2010 at 3:22 PM, Joseph Crivello <josephcrivello at gmail.com>wrote:

>  I think I confused the situation with my ending comment in my last email.
>
>
>
>
> The call is bridged and in the CS_EXCHANGE_MEDIA state when I run the
> transfer.
>
>
>
> What I meant to say in my ending comment in my last email was that I
> noticed if I run the transfer command as listed below when the B-leg is not
> yet in the CS_EXCHANGE_MEDIA state, the transfer works (although of course a
> WARNING is outputted to the console complaining that there is no B-leg).
>
>
>
> -Joe
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Monday, May 03, 2010 3:01 PM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Trouble With Transfer and Async Event
> Socket
>
>
>
> well that's because it's what "both" implies, the "current" leg and the one
> it's bridged to.
>
> if you are not bridged anymore, naturally, it won't work.
>
>
>
>
>
> On Mon, May 3, 2010 at 1:20 PM, Joseph Crivello <josephcrivello at gmail.com>
> wrote:
>
> Example command:
>
>
>
> Command: transfer
>
> Arguments: -both ConfXfer-1 XML default
>
>
>
> And here is the referenced dialplan extension:
>
>
>
>     <extension name="ConfXfer">
>
>       <condition field="destination_number" expression="^ConfXfer-(\d+)$">
>
>         <action application="answer"/>
>
>         <action application="conference" data="$1" />
>
>       </condition>
>
>     </extension>
>
>
>
> Interestingly, I recently discovered that the transfer works if I do it
> before the bridge finishes (figured that out by accident).
>
>
>
> -Joe
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Monday, May 03, 2010 1:06 PM
>
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Trouble With Transfer and Async Event
> Socket
>
>
>
> What command are you using to try to transfer it to your conference?
>
>
>
> On Mon, May 3, 2010 at 9:33 AM, Joe Crivello <josephcrivello at gmail.com>
> wrote:
>
> Hello All,
>
> I am currently working on a conference bridge based on FreeSWITCH that
> allows some conference members to dial out to a prompted number.
>
> Currently my implementation binds * in conference.xml to the "transfer"
> action, leading to a dial plan extension that runs an async full outgoing
> event socket. My application serving the event socket does the following:
>
> 1) Verifies the user is allowed to dial out, and writes a record to the
> database.
> 2) Retrieves the number to be dialed with 'play_and_get_digits'
> 3) Bridges the call to the dialed number
> 4) Waits for the a-leg to press star
> 5) Uses 'transfer' to transfer both legs back to the conference
>
> Everything works up to step #5, which does nothing.
>
> There is no NOTICE outputted to the console indicating that a transfer is
> taking place and neither leg is rejoined to the conference.
>
> I get a normal event back after the transfer sendmsg:
>
> Event-Name: SOCKET_DATA
> Content-Type: command/reply
> Reply-Text: %2BOK
>
> If I subscribe to all events before running the transfer and receive events
> in a loop, I receive nothing but routine RE_SCHEDULE and CALL_UPDATE events
> after the transfer.
>
> I am using ESL to communicate with FreeSWITCH, and prior to running any of
> steps #1-5 I set the event lock to true. The line from the dial plan
> extension that calls my application follows:
>
> <action application="socket" data="172.18.11.1:8001 async full" />
>
> Any ideas? Perhaps I'm "doing it wrong"? I am very new to FS.
>
> Thanks!
> Joe Crivello
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <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
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From aep.lists at it46.se  Mon May  3 13:55:17 2010
From: aep.lists at it46.se (Alberto Escudero)
Date: Mon, 3 May 2010 22:55:17 +0200
Subject: [Freeswitch-users] wrong ptimes and second SDP!,
 cough cough Thomson ST2030
In-Reply-To: <60E86097-89E6-4D9A-829B-C81FE48C6E35@freeswitch.org>
References: <b57c607101b188410fac1458fd0f4edf.squirrel@correo.nodo50.org>
	<31AF5567-8AE7-4ADD-A611-804C6BC6C58E@freeswitch.org>
	<06bafa16896d6e54cc1d5c4e8e5a44bd.squirrel@correo.nodo50.org>
	<60E86097-89E6-4D9A-829B-C81FE48C6E35@freeswitch.org>
Message-ID: <7d267ca45fdacef551ca74a4cb52d900.squirrel@correo.nodo50.org>

I managed to get it working by forcing the phone to use G.711 with 20 ms
and not 10 ms. But for the sake of understanding what exactly

 <param name="inbound-codec-negotiation" value="scrooge"/>
and
 <param name="rtp-autofix-timing" value="true"/>
does?

I saw a BUG in Jira the refers to this second SDP for Sonus
 http://jira.freeswitch.org/browse/FSRTP-8
I wonder if this is Sonus specific patch.

Unfortunately we do not have 16 oz Hammers here in Sweden... so i
recommend this alternative!
http://www.ikea.com/gb/en/catalog/products/70082653





-- 
Stopping junk mailers is good for the environment

> Might I recommend the Craftsman 16 oz. Rip Claw Hammer?
>
> /b
>
> On May 3, 2010, at 1:59 PM, Alberto Escudero wrote:
>
>> I set in internal.xml
>>
>> <param name="inbound-codec-negotiation" value="scrooge"/>
>>
>> But the second SDP returns the ptime to the wrong time.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>




From vfclists at googlemail.com  Mon May  3 14:03:16 2010
From: vfclists at googlemail.com (Frank Church)
Date: Mon, 3 May 2010 22:03:16 +0100
Subject: [Freeswitch-users] mod_xml_cdr error log is not working
In-Reply-To: <68DF5166-42A6-4316-9F96-3FD3CC6C9B62@jerris.com>
References: <g2p7abab2411004291856s2c2d5018g103f33cf2be9dd4b@mail.gmail.com>
	<DD0949EC-0C93-4A8F-9EA4-C6A669AE93FF@jerris.com>
	<l2z7abab2411005030343xdbe1827coc7e531214aa78864@mail.gmail.com>
	<68DF5166-42A6-4316-9F96-3FD3CC6C9B62@jerris.com>
Message-ID: <v2w7abab2411005031403v5e00a8ccx7f2f2a5dadef3b29@mail.gmail.com>

I think it is a bug because the wiki states that log-http-and-disk Default
behaviour is to write either HTTP or Disk on HTTP failure. Setting this to
true will write to both HTTP and Disk regardless (handy for realtime +
reconciliation later if required) true
But it wasn't writing the HTTP connection errors to the disk until I added
log-http-and-disk to the configuration. log-http-and-disk is for writing the
CDR itself, but logging HTTP errors did not work until it was added

http://lists.freeswitch.org/pipermail/freeswitch-users/2010-May/057475.html

On 3 May 2010 21:14, Michael Jerris <mike at jerris.com> wrote:

> would you mind making sure this gets documented on the wiki, and, if it is
> not in the sample configuration, send us a patch for that via
> http://jira.freeswitch.org ?
>
> Mike
>
> On May 3, 2010, at 6:43 AM, Frank Church wrote:
>
> I have received some help about it in
> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-May/057453.html
> .
>
> It requires an additional undocumented parameter in mod_xml_cdr.conf,
> though that may not have been necessary in earlier versions.
>
>
> On 3 May 2010 06:06, Michael Jerris <mike at jerris.com> wrote:
>
>> the defaults in windows are all relative to the running freeswitch base
>> dir, unless you explicitly set them.  there may be some issue with trailing
>> vs no trailing path seperator when you configure them.
>>
>> Mike
>>
>> On Apr 29, 2010, at 9:56 PM, Frank Church wrote:
>>
>> > I have set the err-log-dir and log-dr in xml_cdr.conf.xml but the logs
>> > are not working. The system is a windows system and I wonder if the
>> > defaults for windows are different.
>> >
>> > I have logs/xml_cdr  in addition to the  log/xml_cdr in the
>> > c:\freeswitch directory but Freeswitch can't find them.
>> >
>> > Logs snippet
>> > =========
>> > 2010-04-30 02:41:02.984375 [DEBUG] switch_core_state_machine.c:314
>> > (sofia/internal/1811 at 192.168.1.133) Running State Change CS_REPORTING
>> > 2010-04-30 02:41:02.984375 [DEBUG] switch_core_state_machine.c:590
>> > (sofia/internal/1811 at 192.168.1.133) State REPORTING
>> > 2010-04-30 02:41:03.812500 [ERR] mod_xml_cdr.c:359 Got error [0]
>> > posting to web server [http://192.168.1.20:8132/]
>> > 2010-04-30 02:41:03.812500 [ERR] mod_xml_cdr.c:366 Retry will be with
>> > url [http://192.168.1.20:8132/]
>> > 2010-04-30 02:41:03.812500 [ERR] mod_xml_cdr.c:377 Unable to post to
>> > web server, writing to file
>> > 2010-04-30 02:41:03.812500 [ERR] mod_xml_cdr.c:399 Error![No such file
>> > or directory]
>> >
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Frank Church

=======================
http://devblog.brahmancreations.com
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From vfclists at googlemail.com  Mon May  3 14:05:36 2010
From: vfclists at googlemail.com (Frank Church)
Date: Mon, 3 May 2010 22:05:36 +0100
Subject: [Freeswitch-users] mod_xml_cdr error log is not working
In-Reply-To: <v2w7abab2411005031403v5e00a8ccx7f2f2a5dadef3b29@mail.gmail.com>
References: <g2p7abab2411004291856s2c2d5018g103f33cf2be9dd4b@mail.gmail.com>
	<DD0949EC-0C93-4A8F-9EA4-C6A669AE93FF@jerris.com>
	<l2z7abab2411005030343xdbe1827coc7e531214aa78864@mail.gmail.com>
	<68DF5166-42A6-4316-9F96-3FD3CC6C9B62@jerris.com>
	<v2w7abab2411005031403v5e00a8ccx7f2f2a5dadef3b29@mail.gmail.com>
Message-ID: <i2p7abab2411005031405o7438559bu9bc450aca3f93e35@mail.gmail.com>

PS. Brian has updated the config samples,
http://lists.freeswitch.org/pipermail/freeswitch-users/2010-May/057475.html

On 3 May 2010 22:03, Frank Church <vfclists at googlemail.com> wrote:

>
>
> I think it is a bug because the wiki states that log-http-and-disk Default
> behaviour is to write either HTTP or Disk on HTTP failure. Setting this to
> true will write to both HTTP and Disk regardless (handy for realtime +
> reconciliation later if required) true
> But it wasn't writing the HTTP connection errors to the disk until I added
> log-http-and-disk to the configuration. log-http-and-disk is for writing the
> CDR itself, but logging HTTP errors did not work until it was added
>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-May/057475.html
>
> On 3 May 2010 21:14, Michael Jerris <mike at jerris.com> wrote:
>
>> would you mind making sure this gets documented on the wiki, and, if it is
>> not in the sample configuration, send us a patch for that via
>> http://jira.freeswitch.org ?
>>
>> Mike
>>
>> On May 3, 2010, at 6:43 AM, Frank Church wrote:
>>
>> I have received some help about it in
>> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-May/057453.html
>> .
>>
>> It requires an additional undocumented parameter in mod_xml_cdr.conf,
>> though that may not have been necessary in earlier versions.
>>
>>
>> On 3 May 2010 06:06, Michael Jerris <mike at jerris.com> wrote:
>>
>>> the defaults in windows are all relative to the running freeswitch base
>>> dir, unless you explicitly set them.  there may be some issue with trailing
>>> vs no trailing path seperator when you configure them.
>>>
>>> Mike
>>>
>>> On Apr 29, 2010, at 9:56 PM, Frank Church wrote:
>>>
>>> > I have set the err-log-dir and log-dr in xml_cdr.conf.xml but the logs
>>> > are not working. The system is a windows system and I wonder if the
>>> > defaults for windows are different.
>>> >
>>> > I have logs/xml_cdr  in addition to the  log/xml_cdr in the
>>> > c:\freeswitch directory but Freeswitch can't find them.
>>> >
>>> > Logs snippet
>>> > =========
>>> > 2010-04-30 02:41:02.984375 [DEBUG] switch_core_state_machine.c:314
>>> > (sofia/internal/1811 at 192.168.1.133) Running State Change CS_REPORTING
>>> > 2010-04-30 02:41:02.984375 [DEBUG] switch_core_state_machine.c:590
>>> > (sofia/internal/1811 at 192.168.1.133) State REPORTING
>>> > 2010-04-30 02:41:03.812500 [ERR] mod_xml_cdr.c:359 Got error [0]
>>> > posting to web server [http://192.168.1.20:8132/]
>>> > 2010-04-30 02:41:03.812500 [ERR] mod_xml_cdr.c:366 Retry will be with
>>> > url [http://192.168.1.20:8132/]
>>> > 2010-04-30 02:41:03.812500 [ERR] mod_xml_cdr.c:377 Unable to post to
>>> > web server, writing to file
>>> > 2010-04-30 02:41:03.812500 [ERR] mod_xml_cdr.c:399 Error![No such file
>>> > or directory]
>>> >
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Frank Church
>
> =======================
> http://devblog.brahmancreations.com
>



-- 
Frank Church

=======================
http://devblog.brahmancreations.com
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From peter.olsson at visionutveckling.se  Mon May  3 14:06:09 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 3 May 2010 23:06:09 +0200
Subject: [Freeswitch-users] Suggestions for creating diff's in git...
In-Reply-To: <8E40D6E4-05BE-49D4-8BED-BECA792D6B7A@freeswitch.org>
References: <549CFEF87AEDE841A38E9D15EAB4C04C5576CE4D98@cooper>,
	<8E40D6E4-05BE-49D4-8BED-BECA792D6B7A@freeswitch.org>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5576CE4D99@cooper>

Thanks,

Though, my problem is that I mostly work on Windows, and when piping the output to a file I think it messes up LF/CRLF. I thought if there was another way of doing it - since I can't find a way to output it directly to a file, without messing up CRLF.

I guess I have to start using Linux more :)

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Brian West [brian at freeswitch.org]
Skickat: den 3 maj 2010 22:50
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Suggestions for creating diff's in git...

git diff usually works well.

/b

On May 3, 2010, at 3:43 PM, Peter Olsson wrote:

> I've been using SVN for a couple of years, but I'm quite new to git (as new as the FS project).
>
> I haven't yet found a good (easy) way to create diff's, to send up to jira cases, does anyone have a good tutorial for this?
>
> I've made some patches, but I haven't submitted them yet because of this.
>
> Regards,
>
> Peter


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

!DSPAM:4bdf388532937816820997!



From gchen00 at insightbb.com  Mon May  3 11:02:30 2010
From: gchen00 at insightbb.com (Gary Chen)
Date: Mon, 03 May 2010 14:02:30 -0400
Subject: [Freeswitch-users] Cisco 7960 has problem with music on hold
Message-ID: <fc7dbe899c07.4bded776@insightbb.com>

I just installed FS and registered two cisco 7960 with default configuration file. They can talk to each other.  After connecting both phones, I can put one on hold with music, but I can not get the connection back by pressing resume softkey button. Once I pressing resume button, the connection was hangup.

I can see the cisco 7960 sent INVITE to freeswith once I press resume button but FS sent back 'SIP/2.0 500 Overlapping Requests ' message. This cause cisco 7960 sent out BYE message.

Any idea? 
 
Gary
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From douga at cachecomm.com  Mon May  3 14:39:07 2010
From: douga at cachecomm.com (Doug Albrechtsen)
Date: Mon, 03 May 2010 15:39:07 -0600
Subject: [Freeswitch-users] Caller ID Name Display on Phone
Message-ID: <4BDF427B.8000308@cachecomm.com>

We are using FS 1.0 (svn 17188) with Libpri/dahdi/openzap on Sangoma T-1
card.

When a call comes in on the PRI the Calling Number is shown twice on the
phone.  Both where the number should be and also in place of the Calling
Party Name. Call completes and talks just fine.

A PRI trace on the FS box shows that the name is being received from the
provider in the facility message.
Running the "Info" command in the dialplan shows both caller_id_number
and caller_id_name containing the Callers phone number.

Should Caller-ID Name and Number be shown by default or is there a
setting that needs to be made to show the Name on the phone?

Thanks



From brian at freeswitch.org  Mon May  3 14:48:08 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 3 May 2010 16:48:08 -0500
Subject: [Freeswitch-users] Cisco 7960 has problem with music on hold
In-Reply-To: <fc7dbe899c07.4bded776@insightbb.com>
References: <fc7dbe899c07.4bded776@insightbb.com>
Message-ID: <A90C037C-993E-44F9-A9BA-E95B080E94E2@freeswitch.org>

Well you seem to have left out some details.

1. SIP Load on the 7960?
2. What Rev of FreeSWITCH?
3. No sip traces.

I know it works on my 7960, 7975, 7965 and others.

/b

On May 3, 2010, at 1:02 PM, Gary Chen wrote:

> I just installed FS and registered two cisco 7960 with default configuration file. They can talk to each other.  After connecting both phones, I can put one on hold with music, but I can not get the connection back by pressing resume softkey button. Once I pressing resume button, the connection was hangup.
>  
> I can see the cisco 7960 sent INVITE to freeswith once I press resume button but FS sent back 'SIP/2.0 500 Overlapping Requests ' message. This cause cisco 7960 sent out BYE message.
>  
> Any idea?
>  
> Gary
> ________



From anthony.minessale at gmail.com  Mon May  3 14:57:08 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 3 May 2010 16:57:08 -0500
Subject: [Freeswitch-users] Caller ID Name Display on Phone
In-Reply-To: <4BDF427B.8000308@cachecomm.com>
References: <4BDF427B.8000308@cachecomm.com>
Message-ID: <AANLkTimR1Z9U_M9csP08YRz94Iv-6S8wT71xB1PBVUUH@mail.gmail.com>

the libpri module for openzap may not be getting the info from the correct
field?
you would have to have a look in the code.


On Mon, May 3, 2010 at 4:39 PM, Doug Albrechtsen <douga at cachecomm.com>wrote:

> We are using FS 1.0 (svn 17188) with Libpri/dahdi/openzap on Sangoma T-1
> card.
>
> When a call comes in on the PRI the Calling Number is shown twice on the
> phone.  Both where the number should be and also in place of the Calling
> Party Name. Call completes and talks just fine.
>
> A PRI trace on the FS box shows that the name is being received from the
> provider in the facility message.
> Running the "Info" command in the dialplan shows both caller_id_number
> and caller_id_name containing the Callers phone number.
>
> Should Caller-ID Name and Number be shown by default or is there a
> setting that needs to be made to show the Name on the phone?
>
> Thanks
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
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
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From anthony.minessale at gmail.com  Mon May  3 14:58:18 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 3 May 2010 16:58:18 -0500
Subject: [Freeswitch-users] Cisco 7960 has problem with music on hold
In-Reply-To: <fc7dbe899c07.4bded776@insightbb.com>
References: <fc7dbe899c07.4bded776@insightbb.com>
Message-ID: <AANLkTinJokEEPK9rQvjx_e-kSLLxb6PGUf4JzsncxoPq@mail.gmail.com>

overlapping requests means you are in some nat situation where the server
cannot get replies from your phone.


On Mon, May 3, 2010 at 1:02 PM, Gary Chen <gchen00 at insightbb.com> wrote:

> I just installed FS and registered two cisco 7960 with default
> configuration file. They can talk to each other.  After connecting both
> phones, I can put one on hold with music, but I can not get the connection
> back by pressing resume softkey button. Once I pressing resume button,
> the connection was hangup.
>
> I can see the cisco 7960 sent INVITE to freeswith once I press
> resume button but FS sent back 'SIP/2.0 500 Overlapping Requests ' message.
> This cause cisco 7960 sent out BYE message.
>
>  Any idea?
>
> Gary
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <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
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From vfclists at googlemail.com  Mon May  3 14:59:06 2010
From: vfclists at googlemail.com (Frank Church)
Date: Mon, 3 May 2010 22:59:06 +0100
Subject: [Freeswitch-users] Suggestions for creating diff's in git...
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C5576CE4D99@cooper>
References: <549CFEF87AEDE841A38E9D15EAB4C04C5576CE4D98@cooper>
	<8E40D6E4-05BE-49D4-8BED-BECA792D6B7A@freeswitch.org>
	<549CFEF87AEDE841A38E9D15EAB4C04C5576CE4D99@cooper>
Message-ID: <v2q7abab2411005031459o36c05f43x4f8ce823a08d9b54@mail.gmail.com>

There are options in git for handling CR/LF between Windows and Linux files.
Search Google for 'git crlf'

On 3 May 2010 22:06, Peter Olsson <peter.olsson at visionutveckling.se> wrote:

> Thanks,
>
> Though, my problem is that I mostly work on Windows, and when piping the
> output to a file I think it messes up LF/CRLF. I thought if there was
> another way of doing it - since I can't find a way to output it directly to
> a file, without messing up CRLF.
>
> I guess I have to start using Linux more :)
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [
> freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Brian West [
> brian at freeswitch.org]
> Skickat: den 3 maj 2010 22:50
> Till: freeswitch-users at lists.freeswitch.org
> ?mne: Re: [Freeswitch-users] Suggestions for creating diff's in git...
>
> git diff usually works well.
>
> /b
>
> On May 3, 2010, at 3:43 PM, Peter Olsson wrote:
>
> > I've been using SVN for a couple of years, but I'm quite new to git (as
> new as the FS project).
> >
> > I haven't yet found a good (easy) way to create diff's, to send up to
> jira cases, does anyone have a good tutorial for this?
> >
> > I've made some patches, but I haven't submitted them yet because of this.
> >
> > Regards,
> >
> > Peter
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> !DSPAM:4bdf388532937816820997!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Frank Church

=======================
http://devblog.brahmancreations.com
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From josephcrivello at gmail.com  Mon May  3 16:09:28 2010
From: josephcrivello at gmail.com (Joseph Crivello)
Date: Mon, 3 May 2010 18:09:28 -0500
Subject: [Freeswitch-users] Trouble With Transfer and Async Event Socket
In-Reply-To: <AANLkTimW88uzO9GDBilI8VEvas1D8m3wFCtpgLJN_H2G@mail.gmail.com>
References: <n2sf1b753861005030733u73a51a2fh4fae23812f5089@mail.gmail.com>	<n2y191c3a031005031105v4cd7ad3by96e59cbfc19b319@mail.gmail.com>	<012d01caeaed$47fc3460$d7f49d20$@com>	<AANLkTikZQllnCkEourWKLCu1iykRBBNzKuQazoBt8SVc@mail.gmail.com>	<01ae01caeafe$5168b5d0$f43a2170$@com>
	<AANLkTimW88uzO9GDBilI8VEvas1D8m3wFCtpgLJN_H2G@mail.gmail.com>
Message-ID: <00ac01caeb15$b33695e0$19a3c1a0$@com>

The uuid_transfer command did the trick for me.

 

I wasn't specifying the UUID explicitly with the "transfer" sendmsg, which
happened to be the first command to be sent after the bridge. Apparently you
must explicitly specify the UUID with sendmsg commands after the bridge
command (which I was not aware of).

 

Thanks!

 

-Joe

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, May 03, 2010 3:52 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Trouble With Transfer and Async Event Socket

 

Your attempt to underscore the area where you are having the problem has
obscured the necessary details to help you.

 

You need to re-describe with more details.

 

you might want to use the api interface on your socket app and use
uuid_transfer -both <uuid of a leg> 

On Mon, May 3, 2010 at 3:22 PM, Joseph Crivello <josephcrivello at gmail.com>
wrote:

I think I confused the situation with my ending comment in my last email. 

 

The call is bridged and in the CS_EXCHANGE_MEDIA state when I run the
transfer.

 

What I meant to say in my ending comment in my last email was that I noticed
if I run the transfer command as listed below when the B-leg is not yet in
the CS_EXCHANGE_MEDIA state, the transfer works (although of course a
WARNING is outputted to the console complaining that there is no B-leg).

 

-Joe

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, May 03, 2010 3:01 PM


To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Trouble With Transfer and Async Event Socket

 

well that's because it's what "both" implies, the "current" leg and the one
it's bridged to.

if you are not bridged anymore, naturally, it won't work.

 

 

On Mon, May 3, 2010 at 1:20 PM, Joseph Crivello <josephcrivello at gmail.com>
wrote:

Example command:

 

Command: transfer

Arguments: -both ConfXfer-1 XML default

 

And here is the referenced dialplan extension:

 

    <extension name="ConfXfer">

      <condition field="destination_number" expression="^ConfXfer-(\d+)$">

        <action application="answer"/>

        <action application="conference" data="$1" />

      </condition>

    </extension>

 

Interestingly, I recently discovered that the transfer works if I do it
before the bridge finishes (figured that out by accident).

 

-Joe

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, May 03, 2010 1:06 PM


To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Trouble With Transfer and Async Event Socket

 

What command are you using to try to transfer it to your conference?

 

On Mon, May 3, 2010 at 9:33 AM, Joe Crivello <josephcrivello at gmail.com>
wrote:

Hello All,

I am currently working on a conference bridge based on FreeSWITCH that
allows some conference members to dial out to a prompted number.

Currently my implementation binds * in conference.xml to the "transfer"
action, leading to a dial plan extension that runs an async full outgoing
event socket. My application serving the event socket does the following:

1) Verifies the user is allowed to dial out, and writes a record to the
database.
2) Retrieves the number to be dialed with 'play_and_get_digits'
3) Bridges the call to the dialed number
4) Waits for the a-leg to press star
5) Uses 'transfer' to transfer both legs back to the conference

Everything works up to step #5, which does nothing.

There is no NOTICE outputted to the console indicating that a transfer is
taking place and neither leg is rejoined to the conference.

I get a normal event back after the transfer sendmsg:

Event-Name: SOCKET_DATA
Content-Type: command/reply
Reply-Text: %2BOK

If I subscribe to all events before running the transfer and receive events
in a loop, I receive nothing but routine RE_SCHEDULE and CALL_UPDATE events
after the transfer.

I am using ESL to communicate with FreeSWITCH, and prior to running any of
steps #1-5 I set the event lock to true. The line from the dial plan
extension that calls my application follows:

<action application="socket" data=" <http://172.18.11.1:8001/>
172.18.11.1:8001 async full" />

Any ideas? Perhaps I'm "doing it wrong"? I am very new to FS.

Thanks!
Joe Crivello 
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




-- 
Anthony Minessale II

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

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

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


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




-- 
Anthony Minessale II

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

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

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


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




-- 
Anthony Minessale II

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

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

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

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From msc at freeswitch.org  Mon May  3 17:27:32 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 3 May 2010 17:27:32 -0700
Subject: [Freeswitch-users] run error after building in vs 2008!
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C5576CE4D97@cooper>
References: <i2v1fdf92721004302253l7dbdf254p1ff2b3884ec5a4d4@mail.gmail.com>
	<l2o367751821005010612j46281f6fg17629c38bcd07f4@mail.gmail.com>
	<n2i1fdf92721005011131o221b6dd4sf483e00613963d5@mail.gmail.com>
	<x2z367751821005011229g3b6c217kbda766daac32a44d@mail.gmail.com>
	<E141A983-9EC5-4C62-8D17-675DC0A67009@jerris.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5576CE4D97@cooper>
Message-ID: <k2y87f2f3b91005031727j1661d31vf76dc19d9e3354e5@mail.gmail.com>

Can you put this in a file that we could download and try? Do you have a web
server somewhere that can serve this up?
-MC

On Mon, May 3, 2010 at 1:36 PM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> I also came across this error today on a Windows 2003 server (same issue as
> XP). For now I've commented out the call to inet_ntop(), but I think the
> code below probably could be used - I Googled it. I'll try to find some time
> to get it working within FS. I think there are still some XP/2003 boxes out
> there, so it should be worth fixing.
>
> /Peter
>
> /* const char *
>  * inet_ntop6(src, dst, size)
>  *      convert IPv6 binary address into presentation (printable) format
>  * author:
>  *      Paul Vixie, 1996.
>  */
>
> static const char *
> inet_ntop6(src, dst, size)
>        const u_char *src;
>        char *dst;
>        size_t size;
> {
>        /*
>         * Note that int32_t and int16_t need only be "at least" large
> enough
>         * to contain a value of the specified size.  On some systems, like
>         * Crays, there is no such thing as an integer variable with 16
> bits.
>         * Keep this in mind if you think this function should have been
> coded
>         * to use pointer overlays.  All the world's not a VAX.
>         */
>        char tmp[sizeof "ffff:ffff:ffff:ffff:ffff:ffff:255.255.255.255"],
> *tp;
>        struct { int base, len; } best, cur;
>        u_int words[IN6ADDRSZ / INT16SZ];
>        int i;
>
>        /*
>         * Preprocess:
>         *      Copy the input (bytewise) array into a wordwise array.
>         *      Find the longest run of 0x00's in src[] for :: shorthanding.
>         */
>        memset(words, '\0', sizeof words);
>        for (i = 0; i < IN6ADDRSZ; i++)
>                words[i / 2] |= (src[i] << ((1 - (i % 2)) << 3));
>        best.base = -1;
>        cur.base = -1;
>        for (i = 0; i < (IN6ADDRSZ / INT16SZ); i++) {
>                if (words[i] == 0) {
>                        if (cur.base == -1)
>                                cur.base = i, cur.len = 1;
>                        else
>                                cur.len++;
>                } else {
>                        if (cur.base != -1) {
>                                if (best.base == -1 || cur.len > best.len)
>                                        best = cur;
>                                cur.base = -1;
>                        }
>                }
>        }
>        if (cur.base != -1) {
>                if (best.base == -1 || cur.len > best.len)
>                        best = cur;
>        }
>        if (best.base != -1 && best.len < 2)
>                best.base = -1;
>
>        /*
>         * Format the result.
>         */
>        tp = tmp;
>        for (i = 0; i < (IN6ADDRSZ / INT16SZ); i++) {
>                /* Are we inside the best run of 0x00's? */
>                if (best.base != -1 && i >= best.base &&
>                    i < (best.base + best.len)) {
>                        if (i == best.base)
>                                *tp++ = ':';
>                        continue;
>                }
>                /* Are we following an initial run of 0x00s or any real hex?
> */
>                if (i != 0)
>                        *tp++ = ':';
>                /* Is this address an encapsulated IPv4? */
>                if (i == 6 && best.base == 0 &&
>                    (best.len == 6 || (best.len == 5 && words[5] ==
> 0xffff))) {
>                        if (!inet_ntop4(src+12, tp, sizeof tmp - (tp -
> tmp)))
>                                return (NULL);
>                        tp += strlen(tp);
>                        break;
>                }
>                tp += SPRINTF((tp, "%x", words[i]));
>        }
>        /* Was it a trailing run of 0x00's? */
>        if (best.base != -1 && (best.base + best.len) == (IN6ADDRSZ /
> INT16SZ))
>                *tp++ = ':';
>        *tp++ = '\0';
>
>        /*
>         * Check for overflow, copy, and we're done.
>         */
>        if ((size_t)(tp - tmp) > size) {
>                errno = ENOSPC;
>                return (NULL);
>        }
>        strcpy(dst, tmp);
>        return (dst);
> }
>
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [
> freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Michael Jerris [
> mike at jerris.com]
> Skickat: den 3 maj 2010 06:04
> Till: freeswitch-users at lists.freeswitch.org
> ?mne: Re: [Freeswitch-users] run error after building in vs 2008!
>
> This was just added in :
>
> commit f1430d521a767a13035d6d8e96686564552194fd
> Author: Anthony Minessale <anthm at freeswitch.org<mailto:
> anthm at freeswitch.org>>
> Date:   Fri Apr 30 15:01:32 2010 -0500
>
>    fix switch_get_addr to work with v6 properly
>
> to fix ipv6 support.  We welcome a patch to restore xp support.
>
> Mike
>
> On May 1, 2010, at 3:29 PM, Phillip Jones wrote:
>
> Oh yeah - looks like that call is not supported in older os like XP.
>
>
> http://social.msdn.microsoft.com/Forums/en-US/vcgeneral/thread/e40465f2-41b7-4243-ad33-15ae9366f4e6
>
> FreeSWITCH does support XP though -
> http://wiki.freeswitch.org/wiki/Users_Guide_Introduction
>
> So I don't know.
>
> On Sat, May 1, 2010 at 2:31 PM, babak yakhchali <
> babak.freeswitch at gmail.com<mailto:babak.freeswitch at gmail.com>> wrote:
> ya I've got that in place
> but as I searched the web they say it's related to windows version (xp
> vista . . .)
>
> !DSPAM:4bde4cca32933312412468!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mark.maly at molcs.org  Mon May  3 17:50:17 2010
From: mark.maly at molcs.org (Mark Maly)
Date: Mon, 3 May 2010 19:50:17 -0500
Subject: [Freeswitch-users] Aastra and SCA
In-Reply-To: <21E6CEC8-59C9-4E2A-92ED-D7B93286C5DC@jerris.com>
References: <021d01cae8d2$fc999420$f5ccbc60$@maly@molcs.org>
	<21E6CEC8-59C9-4E2A-92ED-D7B93286C5DC@jerris.com>
Message-ID: <003601caeb23$c7604bc0$5620e340$@maly@molcs.org>

Mike and Brian,

 

Thanks for the information.  Between when I sent the original and received
your responses, I found your Jira entry and sent something to Aastra.  Not
really intending to get anything from them.  But thanks to both of you!

 

Mark

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Jerris
Sent: Sunday, May 02, 2010 11:18 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Aastra and SCA

 

Aastra has a broken SCA implementation.  We have been working with them to
resolve it but as of yet they have failed to deliver a working firmware for
their phones.

 

Mike

 

On Apr 30, 2010, at 10:07 PM, Mark Maly wrote:





Hi,

 

I've tried to patiently figure this out by reading the wiki and this list.
Unfortunately, I've been unable to get it right. 

 

I have 2 Aastra 6731is and a 51i and trying to get SCA working.  I'm
experiencing problems similar to the Cisco thread from last month - outgoing
calls implement SCA well.  Incoming calls ring all lines and appearances
work, but when one phone is answered, the line appearances are removed from
the remaining phones.  I am not attempting to use any DNS.

 

My configuration has all three phones plus FS on a local LAN.  Nothing too
fancy.  Each line is configured for Broadsoft SCA and SCA bridging  is
enabled globally for the phones.

 

Trying to update/replace an old phone at my church.

 

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From pjintheusa at gmail.com  Mon May  3 17:53:38 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Mon, 3 May 2010 20:53:38 -0400
Subject: [Freeswitch-users] Getting the IP (domain) of the FS box a phone
	registered with
Message-ID: <g2i367751821005031753u16511e8ez684f47c7e7f8cf0f@mail.gmail.com>

Hi there,

I have a cluster of FS boxes.  OpenSIPS sits in front of the boxes to load
balance.

The internal profile is sharing the same DB via ODBC.

A mobile phone SIP client is registering with FS1, and a call for that SIP
client arrives on FS2. The mobile phone will not accept unsolicited IP
traffic, so FS1 must send the invite.

>From FS2, how do I find the IP (domain) the mobile phone is registered to?
Without iterating through all my domains with sofia_contact? Or doing a look
up in the DB.


Is there a simpler way?

Many thanks

Pj
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From mark.maly at molcs.org  Mon May  3 17:55:48 2010
From: mark.maly at molcs.org (Mark Maly)
Date: Mon, 3 May 2010 19:55:48 -0500
Subject: [Freeswitch-users] Aastra and SCA
In-Reply-To: <2A6C7EC2-913D-417E-ADD6-E3A72C6AA6E6@freeswitch.org>
References: <021d01cae8d2$fc999420$f5ccbc60$@maly@molcs.org>
	<2A6C7EC2-913D-417E-ADD6-E3A72C6AA6E6@freeswitch.org>
Message-ID: <003b01caeb24$895c7b90$9c1572b0$@maly@molcs.org>

Brian,

 

After my first reply I did receive a msg from Aastra support and whether FS had reported the problem.   The contact wondered whether they [FS] had been given any ?reference number? related to the problem.  

 

Would love to help and would pass ref num if you had one.

 

Mark

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: Sunday, May 02, 2010 11:14 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Aastra and SCA

 

Give up their exists a bug that prevents it from working on any aastra.  Polycom and Cisco SPA work flawless.

 

/b

Sent from my iPad


On Apr 30, 2010, at 9:07 PM, "Mark Maly" <mark.maly at molcs.org> wrote:

Hi,

 

I?ve tried to patiently figure this out by reading the wiki and this list.  Unfortunately, I?ve been unable to get it right.  

 

I have 2 Aastra 6731is and a 51i and trying to get SCA working.  I?m experiencing problems similar to the Cisco thread from last month ? outgoing calls implement SCA well.  Incoming calls ring all lines and appearances work, but when one phone is answered, the line appearances are removed from the remaining phones.  I am not attempting to use any DNS.

 

My configuration has all three phones plus FS on a local LAN.  Nothing too fancy.  Each line is configured for Broadsoft SCA and SCA bridging  is enabled globally for the phones.

 

Trying to update/replace an old phone at my church. 

 

Any help would be greatly appreciated.

 

Thanks, Mark

Mark.maly at molcs.org

 

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From brian at freeswitch.org  Mon May  3 18:00:18 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 3 May 2010 20:00:18 -0500
Subject: [Freeswitch-users] Aastra and SCA
In-Reply-To: <003b01caeb24$895c7b90$9c1572b0$@maly@molcs.org>
References: <021d01cae8d2$fc999420$f5ccbc60$@maly@molcs.org>
	<2A6C7EC2-913D-417E-ADD6-E3A72C6AA6E6@freeswitch.org>
	<003b01caeb24$895c7b90$9c1572b0$@maly@molcs.org>
Message-ID: <28B636AB-D221-44C5-BDA4-1D12AC44748B@freeswitch.org>

Nope but we have exchanged some emails about it back and forth and some beta firmware where they did half way fix it... but seems you need the call-info header on ALL packets associated with the dialog of that call.  They are still missing a few.

/b

On May 3, 2010, at 7:55 PM, Mark Maly wrote:

> Brian,
>  
> After my first reply I did receive a msg from Aastra support and whether FS had reported the problem.   The contact wondered whether they [FS] had been given any ?reference number? related to the problem. 
>  
> Would love to help and would pass ref num if you had one.
>  
> Mark
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
> Sent: Sunday, May 02, 2010 11:14 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Aastra and SCA
>  
> Give up their exists a bug that prevents it from working on any aastra.  Polycom and Cisco SPA work flawless.
>  
> /b
> 
> Sent from my iPad
> 
> On Apr 30, 2010, at 9:07 PM, "Mark Maly" <mark.maly at molcs.org> wrote:
> 
> Hi,
>  
> I?ve tried to patiently figure this out by reading the wiki and this list.  Unfortunately, I?ve been unable to get it right. 
>  
> I have 2 Aastra 6731is and a 51i and trying to get SCA working.  I?m experiencing problems similar to the Cisco thread from last month ? outgoing calls implement SCA well.  Incoming calls ring all lines and appearances work, but when one phone is answered, the line appearances are removed from the remaining phones.  I am not attempting to use any DNS.
>  
> My configuration has all three phones plus FS on a local LAN.  Nothing too fancy.  Each line is configured for Broadsoft SCA and SCA bridging  is enabled globally for the phones.
>  
> Trying to update/replace an old phone at my church.
>  
> Any help would be greatly appreciated.
>  
> Thanks, Mark
> Mark.maly at molcs.org
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mark.maly at molcs.org  Mon May  3 18:06:50 2010
From: mark.maly at molcs.org (Mark Maly)
Date: Mon, 3 May 2010 20:06:50 -0500
Subject: [Freeswitch-users] Aastra and SCA
In-Reply-To: <28B636AB-D221-44C5-BDA4-1D12AC44748B@freeswitch.org>
References: <021d01cae8d2$fc999420$f5ccbc60$@maly@molcs.org>	<2A6C7EC2-913D-417E-ADD6-E3A72C6AA6E6@freeswitch.org>	<003b01caeb24$895c7b90$9c1572b0$@maly@molcs.org>
	<28B636AB-D221-44C5-BDA4-1D12AC44748B@freeswitch.org>
Message-ID: <005201caeb26$1480cb30$3d826190$@maly@molcs.org>

Thanks, I'll pass it on, as well!

Mark

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Monday, May 03, 2010 8:00 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Aastra and SCA

 

Nope but we have exchanged some emails about it back and forth and some beta
firmware where they did half way fix it... but seems you need the call-info
header on ALL packets associated with the dialog of that call.  They are
still missing a few.

 

/b

 

On May 3, 2010, at 7:55 PM, Mark Maly wrote:





Brian,

 

After my first reply I did receive a msg from Aastra support and whether FS
had reported the problem.   The contact wondered whether they [FS] had been
given any "reference number" related to the problem. 

 

Would love to help and would pass ref num if you had one.

 

Mark

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Sunday, May 02, 2010 11:14 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Aastra and SCA

 

Give up their exists a bug that prevents it from working on any aastra.
Polycom and Cisco SPA work flawless.

 

/b

Sent from my iPad


On Apr 30, 2010, at 9:07 PM, "Mark Maly" <mark.maly at molcs.org> wrote:

Hi,

 

I've tried to patiently figure this out by reading the wiki and this list.
Unfortunately, I've been unable to get it right. 

 

I have 2 Aastra 6731is and a 51i and trying to get SCA working.  I'm
experiencing problems similar to the Cisco thread from last month - outgoing
calls implement SCA well.  Incoming calls ring all lines and appearances
work, but when one phone is answered, the line appearances are removed from
the remaining phones.  I am not attempting to use any DNS.

 

My configuration has all three phones plus FS on a local LAN.  Nothing too
fancy.  Each line is configured for Broadsoft SCA and SCA bridging  is
enabled globally for the phones.

 

Trying to update/replace an old phone at my church.

 

Any help would be greatly appreciated.

 

Thanks, Mark

Mark.maly at molcs.org

 

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

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

 

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From nagalenoj at gmail.com  Mon May  3 21:05:20 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Tue, 4 May 2010 09:35:20 +0530
Subject: [Freeswitch-users] How to monitor find out which events are
	currently monitored, set only a particular group of events?
In-Reply-To: <k2w7abab2411005031105n726c11dx434981234cce49f8@mail.gmail.com>
References: <k2w7abab2411005031105n726c11dx434981234cce49f8@mail.gmail.com>
Message-ID: <j2zfe67a3341005032105z4452c3c1n36a1ccc7e0793213@mail.gmail.com>

 I too didn't find any commands to get the registered events. It will be
helpful, if there is some way.

On Mon, May 3, 2010 at 11:35 PM, Frank Church <vfclists at googlemail.com>wrote:

>
> Is there a way of getting of list the event types that are being monitored
> by your connection to Freeswitch, other than those keeping a record of those
> you've added from your own end?
>
> I want to add some without deleting what is present with the first event
> plain XXXX or  filter Event-Name XXXX command
>
> --
> Frank Church
>
> =======================
> http://devblog.brahmancreations.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Regards,
Nagalenoj H.
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From peter.olsson at visionutveckling.se  Mon May  3 23:05:53 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 4 May 2010 08:05:53 +0200
Subject: [Freeswitch-users] run error after building in vs 2008!
In-Reply-To: <k2y87f2f3b91005031727j1661d31vf76dc19d9e3354e5@mail.gmail.com>
References: <i2v1fdf92721004302253l7dbdf254p1ff2b3884ec5a4d4@mail.gmail.com>
	<l2o367751821005010612j46281f6fg17629c38bcd07f4@mail.gmail.com>
	<n2i1fdf92721005011131o221b6dd4sf483e00613963d5@mail.gmail.com>
	<x2z367751821005011229g3b6c217kbda766daac32a44d@mail.gmail.com>
	<E141A983-9EC5-4C62-8D17-675DC0A67009@jerris.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5576CE4D97@cooper>,
	<k2y87f2f3b91005031727j1661d31vf76dc19d9e3354e5@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5576CE4DA1@cooper>

Thanks for the reply.

I found the file on this location: http://cpansearch.perl.org/src/UMEMOTO/Socket6-0.17/inet_ntop.c.

You can download it directly from there.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Michael Collins [msc at freeswitch.org]
Skickat: den 4 maj 2010 02:27
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] run error after building in vs 2008!

Can you put this in a file that we could download and try? Do you have a web server somewhere that can serve this up?
-MC

On Mon, May 3, 2010 at 1:36 PM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
I also came across this error today on a Windows 2003 server (same issue as XP). For now I've commented out the call to inet_ntop(), but I think the code below probably could be used - I Googled it. I'll try to find some time to get it working within FS. I think there are still some XP/2003 boxes out there, so it should be worth fixing.

/Peter

/* const char *
 * inet_ntop6(src, dst, size)
 *      convert IPv6 binary address into presentation (printable) format
 * author:
 *      Paul Vixie, 1996.
 */

static const char *
inet_ntop6(src, dst, size)
       const u_char *src;
       char *dst;
       size_t size;
{
       /*
        * Note that int32_t and int16_t need only be "at least" large enough
        * to contain a value of the specified size.  On some systems, like
        * Crays, there is no such thing as an integer variable with 16 bits.
        * Keep this in mind if you think this function should have been coded
        * to use pointer overlays.  All the world's not a VAX.
        */
       char tmp[sizeof "ffff:ffff:ffff:ffff:ffff:ffff:255.255.255.255"], *tp;
       struct { int base, len; } best, cur;
       u_int words[IN6ADDRSZ / INT16SZ];
       int i;

       /*
        * Preprocess:
        *      Copy the input (bytewise) array into a wordwise array.
        *      Find the longest run of 0x00's in src[] for :: shorthanding.
        */
       memset(words, '\0', sizeof words);
       for (i = 0; i < IN6ADDRSZ; i++)
               words[i / 2] |= (src[i] << ((1 - (i % 2)) << 3));
       best.base = -1;
       cur.base = -1;
       for (i = 0; i < (IN6ADDRSZ / INT16SZ); i++) {
               if (words[i] == 0) {
                       if (cur.base == -1)
                               cur.base = i, cur.len = 1;
                       else
                               cur.len++;
               } else {
                       if (cur.base != -1) {
                               if (best.base == -1 || cur.len > best.len)
                                       best = cur;
                               cur.base = -1;
                       }
               }
       }
       if (cur.base != -1) {
               if (best.base == -1 || cur.len > best.len)
                       best = cur;
       }
       if (best.base != -1 && best.len < 2)
               best.base = -1;

       /*
        * Format the result.
        */
       tp = tmp;
       for (i = 0; i < (IN6ADDRSZ / INT16SZ); i++) {
               /* Are we inside the best run of 0x00's? */
               if (best.base != -1 && i >= best.base &&
                   i < (best.base + best.len)) {
                       if (i == best.base)
                               *tp++ = ':';
                       continue;
               }
               /* Are we following an initial run of 0x00s or any real hex? */
               if (i != 0)
                       *tp++ = ':';
               /* Is this address an encapsulated IPv4? */
               if (i == 6 && best.base == 0 &&
                   (best.len == 6 || (best.len == 5 && words[5] == 0xffff))) {
                       if (!inet_ntop4(src+12, tp, sizeof tmp - (tp - tmp)))
                               return (NULL);
                       tp += strlen(tp);
                       break;
               }
               tp += SPRINTF((tp, "%x", words[i]));
       }
       /* Was it a trailing run of 0x00's? */
       if (best.base != -1 && (best.base + best.len) == (IN6ADDRSZ / INT16SZ))
               *tp++ = ':';
       *tp++ = '\0';

       /*
        * Check for overflow, copy, and we're done.
        */
       if ((size_t)(tp - tmp) > size) {
               errno = ENOSPC;
               return (NULL);
       }
       strcpy(dst, tmp);
       return (dst);
}

________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] f&#246;r Michael Jerris [mike at jerris.com<mailto:mike at jerris.com>]
Skickat: den 3 maj 2010 06:04
Till: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
?mne: Re: [Freeswitch-users] run error after building in vs 2008!

This was just added in :

commit f1430d521a767a13035d6d8e96686564552194fd
Author: Anthony Minessale <anthm at freeswitch.org<mailto:anthm at freeswitch.org><mailto:anthm at freeswitch.org<mailto:anthm at freeswitch.org>>>
Date:   Fri Apr 30 15:01:32 2010 -0500

   fix switch_get_addr to work with v6 properly

to fix ipv6 support.  We welcome a patch to restore xp support.

Mike

On May 1, 2010, at 3:29 PM, Phillip Jones wrote:

Oh yeah - looks like that call is not supported in older os like XP.

http://social.msdn.microsoft.com/Forums/en-US/vcgeneral/thread/e40465f2-41b7-4243-ad33-15ae9366f4e6

FreeSWITCH does support XP though - http://wiki.freeswitch.org/wiki/Users_Guide_Introduction

So I don't know.

On Sat, May 1, 2010 at 2:31 PM, babak yakhchali <babak.freeswitch at gmail.com<mailto:babak.freeswitch at gmail.com><mailto:babak.freeswitch at gmail.com<mailto:babak.freeswitch at gmail.com>>> wrote:
ya I've got that in place
but as I searched the web they say it's related to windows version (xp vista . . .)



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

!DSPAM:4bdf6c0e32936369919130!


From martin at epbx.cz  Mon May  3 23:20:31 2010
From: martin at epbx.cz (Martin Dvorak)
Date: Tue, 04 May 2010 08:20:31 +0200
Subject: [Freeswitch-users] Getting the IP (domain) of the FS box a
 phone registered with
In-Reply-To: <g2i367751821005031753u16511e8ez684f47c7e7f8cf0f@mail.gmail.com>
References: <g2i367751821005031753u16511e8ez684f47c7e7f8cf0f@mail.gmail.com>
Message-ID: <4BDFBCAF.2020108@epbx.cz>

Dne 4.5.2010 2:53, Phillip Jones napsal(a):
> Hi there,
> 
> I have a cluster of FS boxes.  OpenSIPS sits in front of the boxes to
> load balance.
> 
> The internal profile is sharing the same DB via ODBC.
> 
> A mobile phone SIP client is registering with FS1, and a call for that
> SIP client arrives on FS2. The mobile phone will not accept unsolicited
> IP traffic, so FS1 must send the invite.
> 
> From FS2, how do I find the IP (domain) the mobile phone is registered
> to? Without iterating through all my domains with sofia_contact? Or
> doing a look up in the DB.
> 
> 
> Is there a simpler way?
> 

I think yes :-)

If your phones registering "through" OpenSIPS (OpenSIPS just forwards
REGISTERs to FreeSWITCH boxes), they should accept traffic from OpenSIPS
IP address.
And than you could send calls back to your phones through OpenSIPS too.
"Path" support is needed on FreeSWITCH side, but I hope it works (but I
tested it year ago or more).

Beste regards,

kokoska.rokoska


From infos at madovsky.org  Mon May  3 23:48:29 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 4 May 2010 02:48:29 -0400
Subject: [Freeswitch-users] DTMF events
Message-ID: <CEAC61B68CDA414B80B82DA24CDF88AA@MOBILEE1705>

Hi,

is it possible to use min dtmf at 300 ?
it's only to avoid to record again some dtmf files I did 
for a  phone keypad I currently trying to develop

Thanks

Franck
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From brian at freeswitch.org  Mon May  3 23:51:30 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 4 May 2010 01:51:30 -0500
Subject: [Freeswitch-users] Getting the IP (domain) of the FS box a
	phone registered with
In-Reply-To: <4BDFBCAF.2020108@epbx.cz>
References: <g2i367751821005031753u16511e8ez684f47c7e7f8cf0f@mail.gmail.com>
	<4BDFBCAF.2020108@epbx.cz>
Message-ID: <66CEF6A4-2E4D-477E-8BE5-37033BC3F1FF@freeswitch.org>

We already have path support in FreeSWITCH.

/b


On May 4, 2010, at 1:20 AM, Martin Dvorak <martin at epbx.cz> wrote:

> Dne 4.5.2010 2:53, Phillip Jones napsal(a):
>> Hi there,
>> 
>> I have a cluster of FS boxes.  OpenSIPS sits in front of the boxes to
>> load balance.
>> 
>> The internal profile is sharing the same DB via ODBC.
>> 
>> A mobile phone SIP client is registering with FS1, and a call for that
>> SIP client arrives on FS2. The mobile phone will not accept unsolicited
>> IP traffic, so FS1 must send the invite.
>> 
>> From FS2, how do I find the IP (domain) the mobile phone is registered
>> to? Without iterating through all my domains with sofia_contact? Or
>> doing a look up in the DB.
>> 
>> 
>> Is there a simpler way?
>> 
> 
> I think yes :-)
> 
> If your phones registering "through" OpenSIPS (OpenSIPS just forwards
> REGISTERs to FreeSWITCH boxes), they should accept traffic from OpenSIPS
> IP address.
> And than you could send calls back to your phones through OpenSIPS too.
> "Path" support is needed on FreeSWITCH side, but I hope it works (but I
> tested it year ago or more).
> 
> Beste regards,
> 
> kokoska.rokoska
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From kawarod at laposte.net  Tue May  4 01:12:45 2010
From: kawarod at laposte.net (Rod.)
Date: Tue, 04 May 2010 12:12:45 +0400
Subject: [Freeswitch-users] Registration ODBC feeded
 by	another	registrar proxy
In-Reply-To: <SNT135-w18CD702457BE5E08B7D51D2F20@phx.gbl>
References: <4BDED22F.20205@laposte.net>,
	<C5D5A056-B21C-4343-AA13-95D67B3EADA6@gmail.com>,
	<4BDF02AE.3010003@laposte.net>
	<SNT135-w18CD702457BE5E08B7D51D2F20@phx.gbl>
Message-ID: <4BDFD6FD.2050903@laposte.net>

Hi,

I already thought about that, as I'm using static IP it could be even 
easier.
But how to check network connectivity issue and reroute call to 
voicemail asap: call progress timeout ??

rod



Le 03/05/2010 21:42, Jan Berger a ?crit :
> May a suggest a change filter developed if this really is needed?
>
> Re-loading everything just in case something has changes is a huge 
> waste of bandwidth and CPU - if you install an intelligent change 
> filter you would be down to a few entries changing.
>
> Jan
>
> ------------------------------------------------------------------------
> Date: Mon, 3 May 2010 21:06:54 +0400
> From: kawarod at laposte.net
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Registration ODBC feeded by another 
> registrar proxy
>
> Hi,
>
> thanks for your answer and just some details to describe what I'm 
> looking for.
> I have to register 25 000 subscribers, no NAT is involved, each 
> equipment has its own IP address.
> These equipments are registering every 60 seconds on our current 
> platform, but I can change this parameter if needed.
> Equipments are ADSL CPE (router), that's why I'm using 60sec cause 
> flapping could happen very often with ADSL if the copper line is 
> crappy. ADSL could be very unpredictable sometimes.
> As I don't want to delay too much forwarding to voicemail if a user is 
> unavailable (network issue), 60 sec was chosen (bandwith is not an 
> issue). But as I told before, I'm open to your suggestions.
>
> To Philip, using a single SIP proxy (opensips/ser...) in front of a FS 
> cluster could be a single point of failure too.
> I think that maybe a solution using DNS SRV to distribute the load 
> across a cluster could do the trick or some kind of LVS (virtual IP 
> shared across many servers)
> XML curl is a good idea too.
> To be honest, clustering is a must to avoid a single point of failure, 
> but FS performance as a SBC are really great even on commodity 
> hardware, more than 100 CallPerSecond with no transcoding. That's why 
> I think that a mix with a SIP registrar and FS (and redundancy) could 
> easily handle my 25 000 subscribers. I did some lab (one or 2 years 
> ago) with Kamailio registering 90 000 users every 60sec (1500 
> Registration per second) without any issues.
> In my network, 25 000 users are not pushing more than 10 CPS and 500 
> simultaneous call. I'm not doing VoIP termination.
>
> At the moment, I'm just collecting data/feedback on what could be done 
> as I have some time to work on this project, and if going further I 
> will share the configuration as I did before:
> http://wiki.freeswitch.org/wiki/SBC_Setup (not the best setup, but 
> hope it helps users to begin with FS)
>
> regards,
> rod.
>
>
>
>
> Le 03/05/2010 19:54, David Ponzone a ?crit :
>
>     Rod,
>
>     Registering every 60 seconds is a bad idea, and this should not be
>     justified.
>     You should register every 1800 seconds and send a NAT keepalive
>     every X seconds.
>     X should be slightly lower than the NAT UDP timeout of the router
>     in front of the phones, if the phones are behind NAT.
>     If the phones are not behind NAT, NAT keepalive is not necessary.
>
>     David Ponzone Direction Technique
>     email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>     tel:      01 74 03 18 97
>     gsm:   06 66 98 76 34
>
>     Service Client IPeva
>     tel:      0811 46 26 26
>     www.ipeva.fr  - www.ipeva-studio.com
>
>     /Ce message et toutes les pi?ces jointes sont confidentiels et
>     ?tablis ? l'intention exclusive de ses destinataires. Toute
>     utilisation ou diffusion non autoris?e est interdite. Tout message
>     ?lectronique est susceptible d'alt?ration. /*/IPeva/*/ d?cline
>     toute responsabilit? au titre de ce message s'il a ?t? alt?r?,
>     d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>     message, merci de le d?truire imm?diatement et d'avertir
>     l'exp?diteur./
>     /
>     /
>
>
>
>     Le 03/05/2010 ? 15:39, Rod. a ?crit :
>
>         Hi list,
>
>         was playing with FS 1.0.6 and trying to test the registration
>         performance of FS. (Yes I know FS is more suited as a B2BUA,
>         but please
>         read further :p)
>
>         So I did the following:
>             - generate one xml file with 20 000 user account like this:
>         <include>
>         <user id="1">
>         <params>
>         <param name="password" value="1234"/>
>         </params>
>         </user>
>         <user id="2">
>         <params>
>         <param name="password" value="1234"/>
>         </params>
>                     ...
>
>         Then I used Sipp to test how many registration per second
>         could be fired
>         to the server (quad core 2.83Ghz).
>         I setup ulimit variables, and disable nat.
>
>         I got this:
>             - using SQL Lite: unable to get higher than 80
>         registrations per
>         second (in fact it's less than this number but didn't test too
>         much this
>         setup), I see a lot of retransmission in Sipp
>             - using SQL Lite in ramdisk (tmpfs): OK with 80
>         registrations per
>         second but not much
>             - using ODBC and mysql: 130 reg/sec is OK
>
>         With ODBC, above 150 reg/sec I see that FS is stalled to
>         100-110% CPU, I
>         think it's because I'm using only one SIP profile and that
>         SOFIA is
>         monothreaded for this SIP profile.
>         If I'd like to register every 60sec, the server has to support
>         at least
>         more than 300 registration per second.
>
>         So I'm wondering if I could setup something like this:
>             - use another SIP Proxy as a registrar and feed the ODBC
>         "sip_registration database" of FS
>             - FS will be able to use this database to setup a call
>             - use FS as the outbound proxy for call routing
>
>         But what about the user params that have been setup in the xml
>         file
>         above. I think that FS loads the user params each time a user
>         is registered.
>
>         Comments and advices are welcome.
>
>         regards,
>         rod.
>
>
>
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org <http://www.freeswitch.org/>
>
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org  <http://www.freeswitch.org/>
>        
>
>
>
> ------------------------------------------------------------------------
> Hotmail: Free, trusted and rich email service. Get it now. 
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>
>
> _______________________________________________
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>    

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From kawarod at laposte.net  Tue May  4 03:48:18 2010
From: kawarod at laposte.net (Rod.)
Date: Tue, 04 May 2010 14:48:18 +0400
Subject: [Freeswitch-users] wrong ptimes and second SDP!,
 cough cough Thomson ST2030
In-Reply-To: <7d267ca45fdacef551ca74a4cb52d900.squirrel@correo.nodo50.org>
References: <b57c607101b188410fac1458fd0f4edf.squirrel@correo.nodo50.org>	<31AF5567-8AE7-4ADD-A611-804C6BC6C58E@freeswitch.org>	<06bafa16896d6e54cc1d5c4e8e5a44bd.squirrel@correo.nodo50.org>	<60E86097-89E6-4D9A-829B-C81FE48C6E35@freeswitch.org>
	<7d267ca45fdacef551ca74a4cb52d900.squirrel@correo.nodo50.org>
Message-ID: <4BDFFB72.6010406@laposte.net>

Hi,

have you tried an other firmware version. You could find firmware there:
http://www.thomsonbroadbandpartner.com/telephony-solutions/products/product-detail.php?id=87

rod

Le 04/05/2010 00:55, Alberto Escudero a ?crit :
> I managed to get it working by forcing the phone to use G.711 with 20 ms
> and not 10 ms. But for the sake of understanding what exactly
>
>   <param name="inbound-codec-negotiation" value="scrooge"/>
> and
>   <param name="rtp-autofix-timing" value="true"/>
> does?
>
> I saw a BUG in Jira the refers to this second SDP for Sonus
>   http://jira.freeswitch.org/browse/FSRTP-8
> I wonder if this is Sonus specific patch.
>
> Unfortunately we do not have 16 oz Hammers here in Sweden... so i
> recommend this alternative!
> http://www.ikea.com/gb/en/catalog/products/70082653
>
>
>
>
>
>    

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From andy at fabulous4.co.uk  Tue May  4 05:00:11 2010
From: andy at fabulous4.co.uk (Andy)
Date: Tue, 4 May 2010 13:00:11 +0100
Subject: [Freeswitch-users] DTMF stopped working
Message-ID: <9C882B8FF4674A898FF079744EC95629@D810>

Hi folks,
 
What would cause DTMF to suddenly stop working on inbound calls? I have a
relatively simple setup with folks diallg in and navigating through an IVR
menu. I'm using start_dtmf in the dialplan and I can see this being called
at the start of the call. Basically everything was working fine before the
weeked, nothing has changed though we did have some problems with our
internet connection and now none of the dtmf tones in the incoming calls are
being indentified by freeswitch. Any clues? 
 
Many thanks
Andy
 
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From gchen00 at insightbb.com  Tue May  4 05:31:29 2010
From: gchen00 at insightbb.com (Gary Chen)
Date: Tue, 04 May 2010 08:31:29 -0400
Subject: [Freeswitch-users] Cisco 7960 has problem with music on hold
Message-ID: <fc46ab02e9bc.4bdfdb61@insightbb.com>

Here is the detail:
Both Cisco 7960 loaded with SIP firmware: P0S3-08-9-00
They both uses static IP address. 
I am running Freeswitch 1.0.6
The following is the SIP trace: (I altered IP addresses)

U 222.111.000.144:50745 -> 222.111.000.177:5060
  INVITE sip:1004 at 222.111.000.177;user=phone SIP/2.0..Via: SIP/2.0/UDP 222.111.000.144:5060;branch=z9hG4bK43506416..From:
  "Line2" <sip:1008 at 222.111.000.177>;tag=00082166efcb01396b803ac6-7e5ae7dc..To: <sip:1004 at 222.111.000.177;user=phone>..Cal
  l-ID: 00082166-efcb0019-398a09d5-5e799be0 at 222.111.000.144..Max-Forwards: 70..Date: Tue, 04 May 2010 12:17:11 GMT..CSeq:
  101 INVITE..User-Agent: Cisco-CP7960G/8.0..Contact: <sip:1008 at 222.111.000.144:5060;transport=udp>..Expires: 180..Accept:
   application/sdp..Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE..Supported: replaces,join,norefersub
  ..Content-Length: 282..Content-Type: application/sdp..Content-Disposition: session;handling=optional....v=0..o=Cisco-SIP
  UA 11004 0 IN IP4 222.111.000.144..s=SIP Call..t=0 0..m=audio 21654 RTP/AVP 0 8 18 101..c=IN IP4 222.111.000.144..a=rtpm
  ap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101 telephone-event/8000..a=f
  mtp:101 0-15..a=sendrecv..
#
U 222.111.000.177:5060 -> 222.111.000.144:5060
  SIP/2.0 100 Trying..Via: SIP/2.0/UDP 222.111.000.144:5060;branch=z9hG4bK43506416..From: "Line2" <sip:1008 at 222.111.000.17
  7>;tag=00082166efcb01396b803ac6-7e5ae7dc..To: <sip:1004 at 222.111.000.177;user=phone>..Call-ID: 00082166-efcb0019-398a09d5
  -5e799be0 at 222.111.000.144..CSeq: 101 INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported..Content-Length: 0....
#
U 222.111.000.177:5060 -> 222.111.000.102:5060
  INVITE sip:1004 at 222.111.000.102:5060;transport=udp SIP/2.0..Via: SIP/2.0/UDP 222.111.000.177;rport;branch=z9hG4bKB5vj0Sm
  350B9r..Max-Forwards: 68..From: "Line2" <sip:1008 at 222.111.000.177>;tag=9BNeN21mSySQN..To: <sip:1004 at 222.111.000.102:5060
  ;transport=udp>..Call-ID: ff905261-d218-122d-9ab7-6d55cd3bb1e9..CSeq: 130369205 INVITE..Contact: <sip:mod_sofia at 222.111.
  120.177:5060>..User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, U
  PDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Allow-Events:
  talk, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer.
  .Content-Type: application/sdp..Content-Disposition: session..Content-Length: 433..X-FS-Support: update_display..Remote-
  Party-ID: "Line2" <sip:1008 at 222.111.000.177>;party=calling;screen=yes;privacy=off....v=0..o=FreeSWITCH 1272956499 127295
  6500 IN IP4 222.111.000.177..s=FreeSWITCH..c=IN IP4 222.111.000.177..t=0 0..m=audio 18584 RTP/AVP 0 115 107 9 8 3 101 13
  ..a=rtpmap:0 PCMU/8000..a=rtpmap:115 G7221/32000..a=fmtp:115 bitrate=48000..a=rtpmap:107 G7221/16000..a=fmtp:107 bitrate
  =32000..a=rtpmap:9 G722/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:3 GSM/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0
  -16..a=rtpmap:13 CN/8000..a=ptime:20..
#
U 222.111.000.102:51984 -> 222.111.000.177:5060
  SIP/2.0 100 Trying..Via: SIP/2.0/UDP 222.111.000.177;rport;branch=z9hG4bKB5vj0Sm350B9r..From: "Line2" <sip:1008 at 222.111.
  120.177>;tag=9BNeN21mSySQN..To: <sip:1004 at 222.111.000.102:5060;transport=udp>..Call-ID: ff905261-d218-122d-9ab7-6d55cd3b
  b1e9..Date: Tue, 04 May 2010 12:17:12 GMT..CSeq: 130369205 INVITE..Server: Cisco-CP7960G/8.0..Contact: <sip:1001 at 222.111
  .000.102:5060;transport=udp>..Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE..Content-Length: 0....
#
U 222.111.000.102:51985 -> 222.111.000.177:5060
  SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 222.111.000.177;rport;branch=z9hG4bKB5vj0Sm350B9r..From: "Line2" <sip:1008 at 222.111
  .000.177>;tag=9BNeN21mSySQN..To: <sip:1004 at 222.111.000.102:5060;transport=udp>;tag=000821969c8a01964ef2e695-368dc57b..Ca
  ll-ID: ff905261-d218-122d-9ab7-6d55cd3bb1e9..Date: Tue, 04 May 2010 12:17:12 GMT..CSeq: 130369205 INVITE..Server: Cisco-
  CP7960G/8.0..Contact: <sip:1004 at 222.111.000.102:5060;transport=udp>..Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,R
  EGISTER,UPDATE..Content-Length: 0....
#
U 222.111.000.177:5060 -> 222.111.000.144:5060
  SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP 222.111.000.144:5060;branch=z9hG4bK43506416..From: "Line2" <sip:1008 at 204.
  126.000.177>;tag=00082166efcb01396b803ac6-7e5ae7dc..To: <sip:1004 at 222.111.000.177;user=phone>;tag=82UNK7gHvN34S..Call-ID
  : 00082166-efcb0019-398a09d5-5e799be0 at 222.111.000.144..CSeq: 101 INVITE..Contact: <sip:1004 at 222.111.000.177:5060;transpo
  rt=udp>..User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL,
  OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, repla
  ces..Allow-Events: talk, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, mess
  age-summary, refer..Content-Type: application/sdp..Content-Disposition: session..Content-Length: 253..Remote-Party-ID: "
  1004" <sip:1004 at 222.111.000.177>;party=calling;privacy=off;screen=no....v=0..o=FreeSWITCH 1272949011 1272949012 IN IP4 2
  04.126.000.177..s=FreeSWITCH..c=IN IP4 222.111.000.177..t=0 0..m=audio 26072 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpm
  ap:101 telephone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20..
#
U 222.111.000.102:51986 -> 222.111.000.177:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP 222.111.000.177;rport;branch=z9hG4bKB5vj0Sm350B9r..From: "Line2" <sip:1008 at 222.111.000.
  177>;tag=9BNeN21mSySQN..To: <sip:1004 at 222.111.000.102:5060;transport=udp>;tag=000821969c8a01964ef2e695-368dc57b..Call-ID
  : ff905261-d218-122d-9ab7-6d55cd3bb1e9..Date: Tue, 04 May 2010 12:17:13 GMT..CSeq: 130369205 INVITE..Server: Cisco-CP796
  0G/8.0..Contact: <sip:1004 at 222.111.000.102:5060;transport=udp>..Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGIST
  ER,UPDATE..Supported: replaces,join,norefersub..Content-Length: 210..Content-Type: application/sdp..Content-Disposition:
   session;handling=optional....v=0..o=Cisco-SIPUA 5247 0 IN IP4 222.111.000.102..s=SIP Call..t=0 0..m=audio 18454 RTP/AVP
   0 101..c=IN IP4 222.111.000.102..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv..
#
U 222.111.000.177:5060 -> 222.111.000.102:5060
  ACK sip:1004 at 222.111.000.102:5060;transport=udp SIP/2.0..Via: SIP/2.0/UDP 222.111.000.177;rport;branch=z9hG4bKcepB2m5629
  1Um..Max-Forwards: 70..From: "Line2" <sip:1008 at 222.111.000.177>;tag=9BNeN21mSySQN..To: <sip:1004 at 222.111.000.102:5060;tr
  ansport=udp>;tag=000821969c8a01964ef2e695-368dc57b..Call-ID: ff905261-d218-122d-9ab7-6d55cd3bb1e9..CSeq: 130369205 ACK..
  Contact: <sip:mod_sofia at 222.111.000.177:5060>..Content-Length: 0....
#
U 222.111.000.177:5060 -> 222.111.000.144:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP 222.111.000.144:5060;branch=z9hG4bK43506416..From: "Line2" <sip:1008 at 222.111.000.177>;t
  ag=00082166efcb01396b803ac6-7e5ae7dc..To: <sip:1004 at 222.111.000.177;user=phone>;tag=82UNK7gHvN34S..Call-ID: 00082166-efc
  b0019-398a09d5-5e799be0 at 222.111.000.144..CSeq: 101 INVITE..Contact: <sip:1004 at 222.111.000.177:5060;transport=udp>..User-
  Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTE
  R, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Allow-Events: talk, presence, dial
  og, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer..Content-Type: appli
  cation/sdp..Content-Disposition: session..Content-Length: 253..Remote-Party-ID: "1004" <sip:1004 at 222.111.000.177>;party=
  calling;privacy=off;screen=no....v=0..o=FreeSWITCH 1272949011 1272949012 IN IP4 222.111.000.177..s=FreeSWITCH..c=IN IP4
  222.111.000.177..t=0 0..m=audio 26072 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101
   0-16..a=silenceSupp:off - - - -..a=ptime:20..
#
U 222.111.000.144:50746 -> 222.111.000.177:5060
  ACK sip:1004 at 222.111.000.177:5060;transport=udp SIP/2.0..Via: SIP/2.0/UDP 222.111.000.144:5060;branch=z9hG4bK27a3b7e3..F
  rom: "Line2" <sip:1008 at 222.111.000.177>;tag=00082166efcb01396b803ac6-7e5ae7dc..To: <sip:1004 at 222.111.000.177;user=phone>
  ;tag=82UNK7gHvN34S..Call-ID: 00082166-efcb0019-398a09d5-5e799be0 at 222.111.000.144..Max-Forwards: 70..Date: Tue, 04 May 20
  10 12:17:14 GMT..CSeq: 101 ACK..User-Agent: Cisco-CP7960G/8.0..Content-Length: 0....
#
U 222.111.000.102:51018 -> 222.111.000.177:5060
  INVITE sip:mod_sofia at 222.111.000.177:5060 SIP/2.0..Via: SIP/2.0/UDP 222.111.000.102:5060;branch=z9hG4bK6e75964a..From: <
  sip:1004 at 222.111.000.102:5060;transport=udp>;tag=000821969c8a01964ef2e695-368dc57b..To: "Line2" <sip:1008 at 222.111.000.17
  7>;tag=9BNeN21mSySQN..Call-ID: ff905261-d218-122d-9ab7-6d55cd3bb1e9..Max-Forwards: 70..Date: Tue, 04 May 2010 12:17:15 G
  MT..CSeq: 101 INVITE..User-Agent: Cisco-CP7960G/8.0..Contact: <sip:1004 at 222.111.000.102:5060;transport=udp>..Accept: app
  lication/sdp..Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE..Supported: replaces,join,norefersub..Co
  ntent-Length: 281..Content-Type: application/sdp..Content-Disposition: session;handling=optional....v=0..o=Cisco-SIPUA 5
  247 1 IN IP4 222.111.000.102..s=SIP Call..t=0 0..m=audio 18454 RTP/AVP 0 8 18 101..c=IN IP4 222.111.000.102..a=rtpmap:0
  PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101 telephone-event/8000..a=fmtp:1
  01 0-15..a=sendonly..
#
U 222.111.000.177:5060 -> 222.111.000.102:5060
  SIP/2.0 100 Trying..Via: SIP/2.0/UDP 222.111.000.102:5060;branch=z9hG4bK6e75964a..From: <sip:1004 at 222.111.000.102:5060;t
  ransport=udp>;tag=000821969c8a01964ef2e695-368dc57b..To: "Line2" <sip:1008 at 222.111.000.177>;tag=9BNeN21mSySQN..Call-ID:
  ff905261-d218-122d-9ab7-6d55cd3bb1e9..CSeq: 101 INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported..Content-Len
  gth: 0....
#
U 222.111.000.177:5060 -> 222.111.000.102:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP 222.111.000.102:5060;branch=z9hG4bK6e75964a..From: <sip:1004 at 222.111.000.102:5060;trans
  port=udp>;tag=000821969c8a01964ef2e695-368dc57b..To: "Line2" <sip:1008 at 222.111.000.177>;tag=9BNeN21mSySQN..Call-ID: ff90
  5261-d218-122d-9ab7-6d55cd3bb1e9..CSeq: 101 INVITE..Contact: <sip:mod_sofia at 222.111.000.177:5060>..User-Agent: FreeSWITC
  H-mod_sofia/1.0.6-svn-exported..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO
  , REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Content-Type: application
  /sdp..Content-Disposition: session..Content-Length: 265....v=0..o=FreeSWITCH 1272956499 1272956501 IN IP4 222.111.000.17
  7..s=FreeSWITCH..c=IN IP4 222.111.000.177..t=0 0..m=audio 18584 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 teleph
  one-event/8000..a=fmtp:101 0-16..a=recvonly..a=silenceSupp:off - - - -..a=ptime:20..
#
U 222.111.000.177:5060 -> 222.111.000.102:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP 222.111.000.102:5060;branch=z9hG4bK6e75964a..From: <sip:1004 at 222.111.000.102:5060;trans
  port=udp>;tag=000821969c8a01964ef2e695-368dc57b..To: "Line2" <sip:1008 at 222.111.000.177>;tag=9BNeN21mSySQN..Call-ID: ff90
  5261-d218-122d-9ab7-6d55cd3bb1e9..CSeq: 101 INVITE..Contact: <sip:mod_sofia at 222.111.000.177:5060>..User-Agent: FreeSWITC
  H-mod_sofia/1.0.6-svn-exported..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO
  , REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Content-Type: application
  /sdp..Content-Disposition: session..Content-Length: 265....v=0..o=FreeSWITCH 1272956499 1272956501 IN IP4 222.111.000.17
  7..s=FreeSWITCH..c=IN IP4 222.111.000.177..t=0 0..m=audio 18584 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 teleph
  one-event/8000..a=fmtp:101 0-16..a=recvonly..a=silenceSupp:off - - - -..a=ptime:20..
#
U 222.111.000.177:5060 -> 222.111.000.102:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP 222.111.000.102:5060;branch=z9hG4bK6e75964a..From: <sip:1004 at 222.111.000.102:5060;trans
  port=udp>;tag=000821969c8a01964ef2e695-368dc57b..To: "Line2" <sip:1008 at 222.111.000.177>;tag=9BNeN21mSySQN..Call-ID: ff90
  5261-d218-122d-9ab7-6d55cd3bb1e9..CSeq: 101 INVITE..Contact: <sip:mod_sofia at 222.111.000.177:5060>..User-Agent: FreeSWITC
  H-mod_sofia/1.0.6-svn-exported..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO
  , REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Content-Type: application
  /sdp..Content-Disposition: session..Content-Length: 265....v=0..o=FreeSWITCH 1272956499 1272956501 IN IP4 222.111.000.17
  7..s=FreeSWITCH..c=IN IP4 222.111.000.177..t=0 0..m=audio 18584 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 teleph
  one-event/8000..a=fmtp:101 0-16..a=recvonly..a=silenceSupp:off - - - -..a=ptime:20..
#
U 222.111.000.177:5060 -> 222.111.000.102:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP 222.111.000.102:5060;branch=z9hG4bK6e75964a..From: <sip:1004 at 222.111.000.102:5060;trans
  port=udp>;tag=000821969c8a01964ef2e695-368dc57b..To: "Line2" <sip:1008 at 222.111.000.177>;tag=9BNeN21mSySQN..Call-ID: ff90
  5261-d218-122d-9ab7-6d55cd3bb1e9..CSeq: 101 INVITE..Contact: <sip:mod_sofia at 222.111.000.177:5060>..User-Agent: FreeSWITC
  H-mod_sofia/1.0.6-svn-exported..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO
  , REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Content-Type: application
  /sdp..Content-Disposition: session..Content-Length: 265....v=0..o=FreeSWITCH 1272956499 1272956501 IN IP4 222.111.000.17
  7..s=FreeSWITCH..c=IN IP4 222.111.000.177..t=0 0..m=audio 18584 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 teleph
  one-event/8000..a=fmtp:101 0-16..a=recvonly..a=silenceSupp:off - - - -..a=ptime:20..
#
U 222.111.000.102:51018 -> 222.111.000.177:5060
  INVITE sip:mod_sofia at 222.111.000.177:5060 SIP/2.0..Via: SIP/2.0/UDP 222.111.000.102:5060;branch=z9hG4bK2c3b4977..From: <
  sip:1004 at 222.111.000.102:5060;transport=udp>;tag=000821969c8a01964ef2e695-368dc57b..To: "Line2" <sip:1008 at 222.111.000.17
  7>;tag=9BNeN21mSySQN..Call-ID: ff905261-d218-122d-9ab7-6d55cd3bb1e9..Max-Forwards: 70..Date: Tue, 04 May 2010 12:17:22 G
  MT..CSeq: 102 INVITE..User-Agent: Cisco-CP7960G/8.0..Contact: <sip:1004 at 222.111.000.102:5060;transport=udp>..Accept: app
  lication/sdp..Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE..Supported: replaces,join,norefersub..Co
  ntent-Length: 281..Content-Type: application/sdp..Content-Disposition: session;handling=optional....v=0..o=Cisco-SIPUA 5
  247 2 IN IP4 222.111.000.102..s=SIP Call..t=0 0..m=audio 18454 RTP/AVP 0 8 18 101..c=IN IP4 222.111.000.102..a=rtpmap:0
  PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101 telephone-event/8000..a=fmtp:1
  01 0-15..a=sendrecv..
#
U 222.111.000.177:5060 -> 222.111.000.102:5060
  SIP/2.0 500 Overlapping Requests..Via: SIP/2.0/UDP 222.111.000.102:5060;branch=z9hG4bK2c3b4977..From: <sip:1004 at 222.111.
  120.102:5060;transport=udp>;tag=000821969c8a01964ef2e695-368dc57b..To: "Line2" <sip:1008 at 222.111.000.177>;tag=9BNeN21mSy
  SQN..Call-ID: ff905261-d218-122d-9ab7-6d55cd3bb1e9..CSeq: 102 INVITE..Retry-After: 2 (Overlapping Requests)..User-Agent:
   FreeSWITCH-mod_sofia/1.0.6-svn-exported..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UP
  DATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Content-Length:
   0....
#
U 222.111.000.102:51987 -> 222.111.000.177:5060
  ACK sip:mod_sofia at 222.111.000.177:5060 SIP/2.0..Via: SIP/2.0/UDP 222.111.000.102:5060;branch=z9hG4bK2c3b4977..From: <sip
  :1004 at 222.111.000.102:5060;transport=udp>;tag=000821969c8a01964ef2e695-368dc57b..To: "Line2" <sip:1008 at 222.111.000.177>;
  tag=9BNeN21mSySQN..Call-ID: ff905261-d218-122d-9ab7-6d55cd3bb1e9..Date: Tue, 04 May 2010 12:17:23 GMT..CSeq: 102 ACK..Co
  ntent-Length: 0....
#
U 222.111.000.102:51018 -> 222.111.000.177:5060
  BYE sip:mod_sofia at 222.111.000.177:5060 SIP/2.0..Via: SIP/2.0/UDP 222.111.000.102:5060;branch=z9hG4bK7102f98b..From: <sip
  :1004 at 222.111.000.102:5060;transport=udp>;tag=000821969c8a01964ef2e695-368dc57b..To: "Line2" <sip:1008 at 222.111.000.177>;
  tag=9BNeN21mSySQN..Call-ID: ff905261-d218-122d-9ab7-6d55cd3bb1e9..Max-Forwards: 70..Date: Tue, 04 May 2010 12:17:23 GMT.
  .CSeq: 103 BYE..User-Agent: Cisco-CP7960G/8.0..Content-Length: 0....
#
U 222.111.000.177:5060 -> 222.111.000.102:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP 222.111.000.102:5060;branch=z9hG4bK7102f98b..From: <sip:1004 at 222.111.000.102:5060;trans
  port=udp>;tag=000821969c8a01964ef2e695-368dc57b..To: "Line2" <sip:1008 at 222.111.000.177>;tag=9BNeN21mSySQN..Call-ID: ff90
  5261-d218-122d-9ab7-6d55cd3bb1e9..CSeq: 103 BYE..User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported..Allow: INVITE, ACK
  , BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, preconditi
  on, path, replaces..Content-Length: 0....
#
U 222.111.000.177:5060 -> 222.111.000.144:5060
  BYE sip:1008 at 222.111.000.144:5060;transport=udp SIP/2.0..Via: SIP/2.0/UDP 222.111.000.177;rport;branch=z9hG4bKDQF43Fpa0j
  reg..Max-Forwards: 70..From: <sip:1004 at 222.111.000.177;user=phone>;tag=82UNK7gHvN34S..To: "Line2" <sip:1008 at 222.111.000.
  177>;tag=00082166efcb01396b803ac6-7e5ae7dc..Call-ID: 00082166-efcb0019-398a09d5-5e799be0 at 222.111.000.144..CSeq: 13036921
  1 BYE..Contact: <sip:1004 at 222.111.000.177:5060;transport=udp>..User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported..Allo
  w: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: tim
  er, precondition, path, replaces..Reason: Q.850;cause=16;text="NORMAL_CLEARING"..Content-Length: 0....
#
U 222.111.000.144:50747 -> 222.111.000.177:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP 222.111.000.177;rport;branch=z9hG4bKDQF43Fpa0jreg..From: <sip:1004 at 222.111.000.177;user
  =phone>;tag=82UNK7gHvN34S..To: "Line2" <sip:1008 at 222.111.000.177>;tag=00082166efcb01396b803ac6-7e5ae7dc..Call-ID: 000821
  66-efcb0019-398a09d5-5e799be0 at 222.111.000.144..Date: Tue, 04 May 2010 12:17:24 GMT..CSeq: 130369211 BYE..Server: Cisco-C
  P7960G/8.0..Content-Length: 0....
#
U 222.111.000.177:5060 -> 222.111.000.102:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP 222.111.000.102:5060;branch=z9hG4bK6e75964a..From: <sip:1004 at 222.111.000.102:5060;trans
  port=udp>;tag=000821969c8a01964ef2e695-368dc57b..To: "Line2" <sip:1008 at 222.111.000.177>;tag=9BNeN21mSySQN..Call-ID: ff90
  5261-d218-122d-9ab7-6d55cd3bb1e9..CSeq: 101 INVITE..Contact: <sip:mod_sofia at 222.111.000.177:5060>..User-Agent: FreeSWITC
  H-mod_sofia/1.0.6-svn-exported..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO
  , REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Content-Type: application
  /sdp..Content-Disposition: session..Content-Length: 265....v=0..o=FreeSWITCH 1272956499 1272956501 IN IP4 222.111.000.17
  7..s=FreeSWITCH..c=IN IP4 222.111.000.177..t=0 0..m=audio 18584 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 teleph
  one-event/8000..a=fmtp:101 0-16..a=recvonly..a=silenceSupp:off - - - -..a=ptime:20..
#
U 222.111.000.102:51988 -> 222.111.000.177:5060
  ACK sip:mod_sofia at 222.111.000.177:5060 SIP/2.0..Via: SIP/2.0/UDP 222.111.000.102:5060;branch=z9hG4bK2c87fac1..From: <sip
  :1004 at 222.111.000.102:5060;transport=udp>;tag=000821969c8a01964ef2e695-368dc57b..To: "Line2" <sip:1008 at 222.111.000.177>;
  tag=9BNeN21mSySQN..Call-ID: ff905261-d218-122d-9ab7-6d55cd3bb1e9..Max-Forwards: 70..Date: Tue, 04 May 2010 12:17:16 GMT.
  .CSeq: 101 ACK..User-Agent: Cisco-CP7960G/8.0..Content-Length: 0....
exit
25 received, 0 dropped
-------------- next part --------------
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From gchen00 at insightbb.com  Tue May  4 05:35:14 2010
From: gchen00 at insightbb.com (Gary Chen)
Date: Tue, 04 May 2010 08:35:14 -0400
Subject: [Freeswitch-users] Cisco 7960 has problem with music on hold
Message-ID: <fc76c400f578.4bdfdc42@insightbb.com>

 
Here are the details:
Both Cisco 7960 loaded with SIP firmware: P0S3-08-9-00
They both uses static IP address. 
I am running Freeswitch 1.0.6
The following is the SIP trace: (I altered IP addresses)
 
U 222.111.000.144:50745 -> 222.111.000.177:5060
  INVITE sip:1004 at 222.111.000.177;user=phone SIP/2.0..Via: SIP/2.0/UDP 222.111.000.144:5060;branch=z9hG4bK43506416..From:
  "Line2" <sip:1008 at 222.111.000.177>;tag=00082166efcb01396b803ac6-7e5ae7dc..To: <sip:1004 at 222.111.000.177;user=phone>..Cal
  l-ID: 00082166-efcb0019-398a09d5-5e799be0 at 222.111.000.144..Max-Forwards: 70..Date: Tue, 04 May 2010 12:17:11 GMT..CSeq:
  101 INVITE..User-Agent: Cisco-CP7960G/8.0..Contact: <sip:1008 at 222.111.000.144:5060;transport=udp>..Expires: 180..Accept:
   application/sdp..Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE..Supported: replaces,join,norefersub
  ..Content-Length: 282..Content-Type: application/sdp..Content-Disposition: session;handling=optional....v=0..o=Cisco-SIP
  UA 11004 0 IN IP4 222.111.000.144..s=SIP Call..t=0 0..m=audio 21654 RTP/AVP 0 8 18 101..c=IN IP4 222.111.000.144..a=rtpm
  ap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101 telephone-event/8000..a=f
  mtp:101 0-15..a=sendrecv..
#
U 222.111.000.177:5060 -> 222.111.000.144:5060
  SIP/2.0 100 Trying..Via: SIP/2.0/UDP 222.111.000.144:5060;branch=z9hG4bK43506416..From: "Line2" <sip:1008 at 222.111.000.17
  7>;tag=00082166efcb01396b803ac6-7e5ae7dc..To: <sip:1004 at 222.111.000.177;user=phone>..Call-ID: 00082166-efcb0019-398a09d5
  -5e799be0 at 222.111.000.144..CSeq: 101 INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported..Content-Length: 0....
#
U 222.111.000.177:5060 -> 222.111.000.102:5060
  INVITE sip:1004 at 222.111.000.102:5060;transport=udp SIP/2.0..Via: SIP/2.0/UDP 222.111.000.177;rport;branch=z9hG4bKB5vj0Sm
  350B9r..Max-Forwards: 68..From: "Line2" <sip:1008 at 222.111.000.177>;tag=9BNeN21mSySQN..To: <sip:1004 at 222.111.000.102:5060
  ;transport=udp>..Call-ID: ff905261-d218-122d-9ab7-6d55cd3bb1e9..CSeq: 130369205 INVITE..Contact: <sip:mod_sofia at 222.111.
  120.177:5060>..User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, U
  PDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Allow-Events:
  talk, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer.
  .Content-Type: application/sdp..Content-Disposition: session..Content-Length: 433..X-FS-Support: update_display..Remote-
  Party-ID: "Line2" <sip:1008 at 222.111.000.177>;party=calling;screen=yes;privacy=off....v=0..o=FreeSWITCH 1272956499 127295
  6500 IN IP4 222.111.000.177..s=FreeSWITCH..c=IN IP4 222.111.000.177..t=0 0..m=audio 18584 RTP/AVP 0 115 107 9 8 3 101 13
  ..a=rtpmap:0 PCMU/8000..a=rtpmap:115 G7221/32000..a=fmtp:115 bitrate=48000..a=rtpmap:107 G7221/16000..a=fmtp:107 bitrate
  =32000..a=rtpmap:9 G722/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:3 GSM/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0
  -16..a=rtpmap:13 CN/8000..a=ptime:20..
#
U 222.111.000.102:51984 -> 222.111.000.177:5060
  SIP/2.0 100 Trying..Via: SIP/2.0/UDP 222.111.000.177;rport;branch=z9hG4bKB5vj0Sm350B9r..From: "Line2" <sip:1008 at 222.111.
  120.177>;tag=9BNeN21mSySQN..To: <sip:1004 at 222.111.000.102:5060;transport=udp>..Call-ID: ff905261-d218-122d-9ab7-6d55cd3b
  b1e9..Date: Tue, 04 May 2010 12:17:12 GMT..CSeq: 130369205 INVITE..Server: Cisco-CP7960G/8.0..Contact: <sip:1001 at 222.111
  .000.102:5060;transport=udp>..Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE..Content-Length: 0....
#
U 222.111.000.102:51985 -> 222.111.000.177:5060
  SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 222.111.000.177;rport;branch=z9hG4bKB5vj0Sm350B9r..From: "Line2" <sip:1008 at 222.111
  .000.177>;tag=9BNeN21mSySQN..To: <sip:1004 at 222.111.000.102:5060;transport=udp>;tag=000821969c8a01964ef2e695-368dc57b..Ca
  ll-ID: ff905261-d218-122d-9ab7-6d55cd3bb1e9..Date: Tue, 04 May 2010 12:17:12 GMT..CSeq: 130369205 INVITE..Server: Cisco-
  CP7960G/8.0..Contact: <sip:1004 at 222.111.000.102:5060;transport=udp>..Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,R
  EGISTER,UPDATE..Content-Length: 0....
#
U 222.111.000.177:5060 -> 222.111.000.144:5060
  SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP 222.111.000.144:5060;branch=z9hG4bK43506416..From: "Line2" <sip:1008 at 204.
  126.000.177>;tag=00082166efcb01396b803ac6-7e5ae7dc..To: <sip:1004 at 222.111.000.177;user=phone>;tag=82UNK7gHvN34S..Call-ID
  : 00082166-efcb0019-398a09d5-5e799be0 at 222.111.000.144..CSeq: 101 INVITE..Contact: <sip:1004 at 222.111.000.177:5060;transpo
  rt=udp>..User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL,
  OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, repla
  ces..Allow-Events: talk, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, mess
  age-summary, refer..Content-Type: application/sdp..Content-Disposition: session..Content-Length: 253..Remote-Party-ID: "
  1004" <sip:1004 at 222.111.000.177>;party=calling;privacy=off;screen=no....v=0..o=FreeSWITCH 1272949011 1272949012 IN IP4 2
  04.126.000.177..s=FreeSWITCH..c=IN IP4 222.111.000.177..t=0 0..m=audio 26072 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpm
  ap:101 telephone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20..
#
U 222.111.000.102:51986 -> 222.111.000.177:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP 222.111.000.177;rport;branch=z9hG4bKB5vj0Sm350B9r..From: "Line2" <sip:1008 at 222.111.000.
  177>;tag=9BNeN21mSySQN..To: <sip:1004 at 222.111.000.102:5060;transport=udp>;tag=000821969c8a01964ef2e695-368dc57b..Call-ID
  : ff905261-d218-122d-9ab7-6d55cd3bb1e9..Date: Tue, 04 May 2010 12:17:13 GMT..CSeq: 130369205 INVITE..Server: Cisco-CP796
  0G/8.0..Contact: <sip:1004 at 222.111.000.102:5060;transport=udp>..Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGIST
  ER,UPDATE..Supported: replaces,join,norefersub..Content-Length: 210..Content-Type: application/sdp..Content-Disposition:
   session;handling=optional....v=0..o=Cisco-SIPUA 5247 0 IN IP4 222.111.000.102..s=SIP Call..t=0 0..m=audio 18454 RTP/AVP
   0 101..c=IN IP4 222.111.000.102..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv..
#
U 222.111.000.177:5060 -> 222.111.000.102:5060
  ACK sip:1004 at 222.111.000.102:5060;transport=udp SIP/2.0..Via: SIP/2.0/UDP 222.111.000.177;rport;branch=z9hG4bKcepB2m5629
  1Um..Max-Forwards: 70..From: "Line2" <sip:1008 at 222.111.000.177>;tag=9BNeN21mSySQN..To: <sip:1004 at 222.111.000.102:5060;tr
  ansport=udp>;tag=000821969c8a01964ef2e695-368dc57b..Call-ID: ff905261-d218-122d-9ab7-6d55cd3bb1e9..CSeq: 130369205 ACK..
  Contact: <sip:mod_sofia at 222.111.000.177:5060>..Content-Length: 0....
#
U 222.111.000.177:5060 -> 222.111.000.144:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP 222.111.000.144:5060;branch=z9hG4bK43506416..From: "Line2" <sip:1008 at 222.111.000.177>;t
  ag=00082166efcb01396b803ac6-7e5ae7dc..To: <sip:1004 at 222.111.000.177;user=phone>;tag=82UNK7gHvN34S..Call-ID: 00082166-efc
  b0019-398a09d5-5e799be0 at 222.111.000.144..CSeq: 101 INVITE..Contact: <sip:1004 at 222.111.000.177:5060;transport=udp>..User-
  Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTE
  R, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Allow-Events: talk, presence, dial
  og, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer..Content-Type: appli
  cation/sdp..Content-Disposition: session..Content-Length: 253..Remote-Party-ID: "1004" <sip:1004 at 222.111.000.177>;party=
  calling;privacy=off;screen=no....v=0..o=FreeSWITCH 1272949011 1272949012 IN IP4 222.111.000.177..s=FreeSWITCH..c=IN IP4
  222.111.000.177..t=0 0..m=audio 26072 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101
   0-16..a=silenceSupp:off - - - -..a=ptime:20..
#
U 222.111.000.144:50746 -> 222.111.000.177:5060
  ACK sip:1004 at 222.111.000.177:5060;transport=udp SIP/2.0..Via: SIP/2.0/UDP 222.111.000.144:5060;branch=z9hG4bK27a3b7e3..F
  rom: "Line2" <sip:1008 at 222.111.000.177>;tag=00082166efcb01396b803ac6-7e5ae7dc..To: <sip:1004 at 222.111.000.177;user=phone>
  ;tag=82UNK7gHvN34S..Call-ID: 00082166-efcb0019-398a09d5-5e799be0 at 222.111.000.144..Max-Forwards: 70..Date: Tue, 04 May 20
  10 12:17:14 GMT..CSeq: 101 ACK..User-Agent: Cisco-CP7960G/8.0..Content-Length: 0....
#
U 222.111.000.102:51018 -> 222.111.000.177:5060
  INVITE sip:mod_sofia at 222.111.000.177:5060 SIP/2.0..Via: SIP/2.0/UDP 222.111.000.102:5060;branch=z9hG4bK6e75964a..From: <
  sip:1004 at 222.111.000.102:5060;transport=udp>;tag=000821969c8a01964ef2e695-368dc57b..To: "Line2" <sip:1008 at 222.111.000.17
  7>;tag=9BNeN21mSySQN..Call-ID: ff905261-d218-122d-9ab7-6d55cd3bb1e9..Max-Forwards: 70..Date: Tue, 04 May 2010 12:17:15 G
  MT..CSeq: 101 INVITE..User-Agent: Cisco-CP7960G/8.0..Contact: <sip:1004 at 222.111.000.102:5060;transport=udp>..Accept: app
  lication/sdp..Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE..Supported: replaces,join,norefersub..Co
  ntent-Length: 281..Content-Type: application/sdp..Content-Disposition: session;handling=optional....v=0..o=Cisco-SIPUA 5
  247 1 IN IP4 222.111.000.102..s=SIP Call..t=0 0..m=audio 18454 RTP/AVP 0 8 18 101..c=IN IP4 222.111.000.102..a=rtpmap:0
  PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101 telephone-event/8000..a=fmtp:1
  01 0-15..a=sendonly..
#
U 222.111.000.177:5060 -> 222.111.000.102:5060
  SIP/2.0 100 Trying..Via: SIP/2.0/UDP 222.111.000.102:5060;branch=z9hG4bK6e75964a..From: <sip:1004 at 222.111.000.102:5060;t
  ransport=udp>;tag=000821969c8a01964ef2e695-368dc57b..To: "Line2" <sip:1008 at 222.111.000.177>;tag=9BNeN21mSySQN..Call-ID:
  ff905261-d218-122d-9ab7-6d55cd3bb1e9..CSeq: 101 INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported..Content-Len
  gth: 0....
#
U 222.111.000.177:5060 -> 222.111.000.102:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP 222.111.000.102:5060;branch=z9hG4bK6e75964a..From: <sip:1004 at 222.111.000.102:5060;trans
  port=udp>;tag=000821969c8a01964ef2e695-368dc57b..To: "Line2" <sip:1008 at 222.111.000.177>;tag=9BNeN21mSySQN..Call-ID: ff90
  5261-d218-122d-9ab7-6d55cd3bb1e9..CSeq: 101 INVITE..Contact: <sip:mod_sofia at 222.111.000.177:5060>..User-Agent: FreeSWITC
  H-mod_sofia/1.0.6-svn-exported..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO
  , REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Content-Type: application
  /sdp..Content-Disposition: session..Content-Length: 265....v=0..o=FreeSWITCH 1272956499 1272956501 IN IP4 222.111.000.17
  7..s=FreeSWITCH..c=IN IP4 222.111.000.177..t=0 0..m=audio 18584 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 teleph
  one-event/8000..a=fmtp:101 0-16..a=recvonly..a=silenceSupp:off - - - -..a=ptime:20..
#
U 222.111.000.177:5060 -> 222.111.000.102:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP 222.111.000.102:5060;branch=z9hG4bK6e75964a..From: <sip:1004 at 222.111.000.102:5060;trans
  port=udp>;tag=000821969c8a01964ef2e695-368dc57b..To: "Line2" <sip:1008 at 222.111.000.177>;tag=9BNeN21mSySQN..Call-ID: ff90
  5261-d218-122d-9ab7-6d55cd3bb1e9..CSeq: 101 INVITE..Contact: <sip:mod_sofia at 222.111.000.177:5060>..User-Agent: FreeSWITC
  H-mod_sofia/1.0.6-svn-exported..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO
  , REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Content-Type: application
  /sdp..Content-Disposition: session..Content-Length: 265....v=0..o=FreeSWITCH 1272956499 1272956501 IN IP4 222.111.000.17
  7..s=FreeSWITCH..c=IN IP4 222.111.000.177..t=0 0..m=audio 18584 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 teleph
  one-event/8000..a=fmtp:101 0-16..a=recvonly..a=silenceSupp:off - - - -..a=ptime:20..
#
U 222.111.000.177:5060 -> 222.111.000.102:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP 222.111.000.102:5060;branch=z9hG4bK6e75964a..From: <sip:1004 at 222.111.000.102:5060;trans
  port=udp>;tag=000821969c8a01964ef2e695-368dc57b..To: "Line2" <sip:1008 at 222.111.000.177>;tag=9BNeN21mSySQN..Call-ID: ff90
  5261-d218-122d-9ab7-6d55cd3bb1e9..CSeq: 101 INVITE..Contact: <sip:mod_sofia at 222.111.000.177:5060>..User-Agent: FreeSWITC
  H-mod_sofia/1.0.6-svn-exported..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO
  , REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Content-Type: application
  /sdp..Content-Disposition: session..Content-Length: 265....v=0..o=FreeSWITCH 1272956499 1272956501 IN IP4 222.111.000.17
  7..s=FreeSWITCH..c=IN IP4 222.111.000.177..t=0 0..m=audio 18584 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 teleph
  one-event/8000..a=fmtp:101 0-16..a=recvonly..a=silenceSupp:off - - - -..a=ptime:20..
#
U 222.111.000.177:5060 -> 222.111.000.102:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP 222.111.000.102:5060;branch=z9hG4bK6e75964a..From: <sip:1004 at 222.111.000.102:5060;trans
  port=udp>;tag=000821969c8a01964ef2e695-368dc57b..To: "Line2" <sip:1008 at 222.111.000.177>;tag=9BNeN21mSySQN..Call-ID: ff90
  5261-d218-122d-9ab7-6d55cd3bb1e9..CSeq: 101 INVITE..Contact: <sip:mod_sofia at 222.111.000.177:5060>..User-Agent: FreeSWITC
  H-mod_sofia/1.0.6-svn-exported..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO
  , REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Content-Type: application
  /sdp..Content-Disposition: session..Content-Length: 265....v=0..o=FreeSWITCH 1272956499 1272956501 IN IP4 222.111.000.17
  7..s=FreeSWITCH..c=IN IP4 222.111.000.177..t=0 0..m=audio 18584 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 teleph
  one-event/8000..a=fmtp:101 0-16..a=recvonly..a=silenceSupp:off - - - -..a=ptime:20..
#
U 222.111.000.102:51018 -> 222.111.000.177:5060
  INVITE sip:mod_sofia at 222.111.000.177:5060 SIP/2.0..Via: SIP/2.0/UDP 222.111.000.102:5060;branch=z9hG4bK2c3b4977..From: <
  sip:1004 at 222.111.000.102:5060;transport=udp>;tag=000821969c8a01964ef2e695-368dc57b..To: "Line2" <sip:1008 at 222.111.000.17
  7>;tag=9BNeN21mSySQN..Call-ID: ff905261-d218-122d-9ab7-6d55cd3bb1e9..Max-Forwards: 70..Date: Tue, 04 May 2010 12:17:22 G
  MT..CSeq: 102 INVITE..User-Agent: Cisco-CP7960G/8.0..Contact: <sip:1004 at 222.111.000.102:5060;transport=udp>..Accept: app
  lication/sdp..Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE..Supported: replaces,join,norefersub..Co
  ntent-Length: 281..Content-Type: application/sdp..Content-Disposition: session;handling=optional....v=0..o=Cisco-SIPUA 5
  247 2 IN IP4 222.111.000.102..s=SIP Call..t=0 0..m=audio 18454 RTP/AVP 0 8 18 101..c=IN IP4 222.111.000.102..a=rtpmap:0
  PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101 telephone-event/8000..a=fmtp:1
  01 0-15..a=sendrecv..
#
U 222.111.000.177:5060 -> 222.111.000.102:5060
  SIP/2.0 500 Overlapping Requests..Via: SIP/2.0/UDP 222.111.000.102:5060;branch=z9hG4bK2c3b4977..From: <sip:1004 at 222.111.
  120.102:5060;transport=udp>;tag=000821969c8a01964ef2e695-368dc57b..To: "Line2" <sip:1008 at 222.111.000.177>;tag=9BNeN21mSy
  SQN..Call-ID: ff905261-d218-122d-9ab7-6d55cd3bb1e9..CSeq: 102 INVITE..Retry-After: 2 (Overlapping Requests)..User-Agent:
   FreeSWITCH-mod_sofia/1.0.6-svn-exported..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UP
  DATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Content-Length:
   0....
#
U 222.111.000.102:51987 -> 222.111.000.177:5060
  ACK sip:mod_sofia at 222.111.000.177:5060 SIP/2.0..Via: SIP/2.0/UDP 222.111.000.102:5060;branch=z9hG4bK2c3b4977..From: <sip
  :1004 at 222.111.000.102:5060;transport=udp>;tag=000821969c8a01964ef2e695-368dc57b..To: "Line2" <sip:1008 at 222.111.000.177>;
  tag=9BNeN21mSySQN..Call-ID: ff905261-d218-122d-9ab7-6d55cd3bb1e9..Date: Tue, 04 May 2010 12:17:23 GMT..CSeq: 102 ACK..Co
  ntent-Length: 0....
#
U 222.111.000.102:51018 -> 222.111.000.177:5060
  BYE sip:mod_sofia at 222.111.000.177:5060 SIP/2.0..Via: SIP/2.0/UDP 222.111.000.102:5060;branch=z9hG4bK7102f98b..From: <sip
  :1004 at 222.111.000.102:5060;transport=udp>;tag=000821969c8a01964ef2e695-368dc57b..To: "Line2" <sip:1008 at 222.111.000.177>;
  tag=9BNeN21mSySQN..Call-ID: ff905261-d218-122d-9ab7-6d55cd3bb1e9..Max-Forwards: 70..Date: Tue, 04 May 2010 12:17:23 GMT.
  .CSeq: 103 BYE..User-Agent: Cisco-CP7960G/8.0..Content-Length: 0....
#
U 222.111.000.177:5060 -> 222.111.000.102:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP 222.111.000.102:5060;branch=z9hG4bK7102f98b..From: <sip:1004 at 222.111.000.102:5060;trans
  port=udp>;tag=000821969c8a01964ef2e695-368dc57b..To: "Line2" <sip:1008 at 222.111.000.177>;tag=9BNeN21mSySQN..Call-ID: ff90
  5261-d218-122d-9ab7-6d55cd3bb1e9..CSeq: 103 BYE..User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported..Allow: INVITE, ACK
  , BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, preconditi
  on, path, replaces..Content-Length: 0....
#
U 222.111.000.177:5060 -> 222.111.000.144:5060
  BYE sip:1008 at 222.111.000.144:5060;transport=udp SIP/2.0..Via: SIP/2.0/UDP 222.111.000.177;rport;branch=z9hG4bKDQF43Fpa0j
  reg..Max-Forwards: 70..From: <sip:1004 at 222.111.000.177;user=phone>;tag=82UNK7gHvN34S..To: "Line2" <sip:1008 at 222.111.000.
  177>;tag=00082166efcb01396b803ac6-7e5ae7dc..Call-ID: 00082166-efcb0019-398a09d5-5e799be0 at 222.111.000.144..CSeq: 13036921
  1 BYE..Contact: <sip:1004 at 222.111.000.177:5060;transport=udp>..User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported..Allo
  w: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: tim
  er, precondition, path, replaces..Reason: Q.850;cause=16;text="NORMAL_CLEARING"..Content-Length: 0....
#
U 222.111.000.144:50747 -> 222.111.000.177:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP 222.111.000.177;rport;branch=z9hG4bKDQF43Fpa0jreg..From: <sip:1004 at 222.111.000.177;user
  =phone>;tag=82UNK7gHvN34S..To: "Line2" <sip:1008 at 222.111.000.177>;tag=00082166efcb01396b803ac6-7e5ae7dc..Call-ID: 000821
  66-efcb0019-398a09d5-5e799be0 at 222.111.000.144..Date: Tue, 04 May 2010 12:17:24 GMT..CSeq: 130369211 BYE..Server: Cisco-C
  P7960G/8.0..Content-Length: 0....
#
U 222.111.000.177:5060 -> 222.111.000.102:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP 222.111.000.102:5060;branch=z9hG4bK6e75964a..From: <sip:1004 at 222.111.000.102:5060;trans
  port=udp>;tag=000821969c8a01964ef2e695-368dc57b..To: "Line2" <sip:1008 at 222.111.000.177>;tag=9BNeN21mSySQN..Call-ID: ff90
  5261-d218-122d-9ab7-6d55cd3bb1e9..CSeq: 101 INVITE..Contact: <sip:mod_sofia at 222.111.000.177:5060>..User-Agent: FreeSWITC
  H-mod_sofia/1.0.6-svn-exported..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO
  , REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Content-Type: application
  /sdp..Content-Disposition: session..Content-Length: 265....v=0..o=FreeSWITCH 1272956499 1272956501 IN IP4 222.111.000.17
  7..s=FreeSWITCH..c=IN IP4 222.111.000.177..t=0 0..m=audio 18584 RTP/AVP 0 101..a=rtpmap:0 PCMU/8000..a=rtpmap:101 teleph
  one-event/8000..a=fmtp:101 0-16..a=recvonly..a=silenceSupp:off - - - -..a=ptime:20..
#
U 222.111.000.102:51988 -> 222.111.000.177:5060
  ACK sip:mod_sofia at 222.111.000.177:5060 SIP/2.0..Via: SIP/2.0/UDP 222.111.000.102:5060;branch=z9hG4bK2c87fac1..From: <sip
  :1004 at 222.111.000.102:5060;transport=udp>;tag=000821969c8a01964ef2e695-368dc57b..To: "Line2" <sip:1008 at 222.111.000.177>;
  tag=9BNeN21mSySQN..Call-ID: ff905261-d218-122d-9ab7-6d55cd3bb1e9..Max-Forwards: 70..Date: Tue, 04 May 2010 12:17:16 GMT.
  .CSeq: 101 ACK..User-Agent: Cisco-CP7960G/8.0..Content-Length: 0....
exit
25 received, 0 dropped


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: Monday, May 03, 2010 5:48 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Cisco 7960 has problem with music on hold
 
Well you seem to have left out some details.
 
1. SIP Load on the 7960?
2. What Rev of FreeSWITCH?
3. No sip traces.
 
I know it works on my 7960, 7975, 7965 and others.
 
/b
 
On May 3, 2010, at 1:02 PM, Gary Chen wrote:
 
> I just installed FS and registered two cisco 7960 with default configuration file. They can talk to each other.  After connecting both phones, I can put one on hold with music, but I can not get the connection back by pressing resume softkey button. Once I pressing resume button, the connection was hangup.
>  
> I can see the cisco 7960 sent INVITE to freeswith once I press resume button but FS sent back 'SIP/2.0 500 Overlapping Requests ' message. This cause cisco 7960 sent out BYE message.
>  
> Any idea?
>  
> Gary
> ________
 
 
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From pjintheusa at gmail.com  Tue May  4 05:38:02 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Tue, 4 May 2010 08:38:02 -0400
Subject: [Freeswitch-users] Getting the IP (domain) of the FS box a
	phone registered with
In-Reply-To: <66CEF6A4-2E4D-477E-8BE5-37033BC3F1FF@freeswitch.org>
References: <g2i367751821005031753u16511e8ez684f47c7e7f8cf0f@mail.gmail.com>
	<4BDFBCAF.2020108@epbx.cz>
	<66CEF6A4-2E4D-477E-8BE5-37033BC3F1FF@freeswitch.org>
Message-ID: <w2s367751821005040538sdfabcbf4wc9b1f5340d98dea7@mail.gmail.com>

kokoska,

thanks for that. I am considering setting up opensips to deal with mobile
phone connections (as well as carrier connections) but it just introduces a
single point of failure. I am trying to ing to avoid this.


At this point I am wondering whether there is a simple way of finding out
which FS box a client is registered with. (without doing a SQL query into
the shared DB - although that would be ok too)

I am wondering then whether I can use FS_PATH to route through that box
(domain)

Thanks!

Pj


On Tue, May 4, 2010 at 2:51 AM, Brian West <brian at freeswitch.org> wrote:

> We already have path support in FreeSWITCH.
>
> /b
>
>
> On May 4, 2010, at 1:20 AM, Martin Dvorak <martin at epbx.cz> wrote:
>
> > Dne 4.5.2010 2:53, Phillip Jones napsal(a):
> >> Hi there,
> >>
> >> I have a cluster of FS boxes.  OpenSIPS sits in front of the boxes to
> >> load balance.
> >>
> >> The internal profile is sharing the same DB via ODBC.
> >>
> >> A mobile phone SIP client is registering with FS1, and a call for that
> >> SIP client arrives on FS2. The mobile phone will not accept unsolicited
> >> IP traffic, so FS1 must send the invite.
> >>
> >> From FS2, how do I find the IP (domain) the mobile phone is registered
> >> to? Without iterating through all my domains with sofia_contact? Or
> >> doing a look up in the DB.
> >>
> >>
> >> Is there a simpler way?
> >>
> >
> > I think yes :-)
> >
> > If your phones registering "through" OpenSIPS (OpenSIPS just forwards
> > REGISTERs to FreeSWITCH boxes), they should accept traffic from OpenSIPS
> > IP address.
> > And than you could send calls back to your phones through OpenSIPS too.
> > "Path" support is needed on FreeSWITCH side, but I hope it works (but I
> > tested it year ago or more).
> >
> > Beste regards,
> >
> > kokoska.rokoska
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From red.rain.seven at gmail.com  Tue May  4 06:04:33 2010
From: red.rain.seven at gmail.com (Henry Huang)
Date: Tue, 4 May 2010 21:04:33 +0800
Subject: [Freeswitch-users] Getting the IP (domain) of the FS box a
	phone registered with
In-Reply-To: <w2s367751821005040538sdfabcbf4wc9b1f5340d98dea7@mail.gmail.com>
References: <g2i367751821005031753u16511e8ez684f47c7e7f8cf0f@mail.gmail.com>
	<4BDFBCAF.2020108@epbx.cz>
	<66CEF6A4-2E4D-477E-8BE5-37033BC3F1FF@freeswitch.org>
	<w2s367751821005040538sdfabcbf4wc9b1f5340d98dea7@mail.gmail.com>
Message-ID: <i2z59ad9ca11005040604kf8ea18dckd99a842fbfc0966c@mail.gmail.com>

Brian:

Where can I find information regarding FreeSWITCH path?

Thanks,

Henry


On Tue, May 4, 2010 at 8:38 PM, Phillip Jones <pjintheusa at gmail.com> wrote:

> kokoska,
>
> thanks for that. I am considering setting up opensips to deal with mobile
> phone connections (as well as carrier connections) but it just introduces a
> single point of failure. I am trying to ing to avoid this.
>
>
> At this point I am wondering whether there is a simple way of finding out
> which FS box a client is registered with. (without doing a SQL query into
> the shared DB - although that would be ok too)
>
> I am wondering then whether I can use FS_PATH to route through that box
> (domain)
>
> Thanks!
>
> Pj
>
>
>
> On Tue, May 4, 2010 at 2:51 AM, Brian West <brian at freeswitch.org> wrote:
>
>> We already have path support in FreeSWITCH.
>>
>> /b
>>
>>
>> On May 4, 2010, at 1:20 AM, Martin Dvorak <martin at epbx.cz> wrote:
>>
>> > Dne 4.5.2010 2:53, Phillip Jones napsal(a):
>> >> Hi there,
>> >>
>> >> I have a cluster of FS boxes.  OpenSIPS sits in front of the boxes to
>> >> load balance.
>> >>
>> >> The internal profile is sharing the same DB via ODBC.
>> >>
>> >> A mobile phone SIP client is registering with FS1, and a call for that
>> >> SIP client arrives on FS2. The mobile phone will not accept unsolicited
>> >> IP traffic, so FS1 must send the invite.
>> >>
>> >> From FS2, how do I find the IP (domain) the mobile phone is registered
>> >> to? Without iterating through all my domains with sofia_contact? Or
>> >> doing a look up in the DB.
>> >>
>> >>
>> >> Is there a simpler way?
>> >>
>> >
>> > I think yes :-)
>> >
>> > If your phones registering "through" OpenSIPS (OpenSIPS just forwards
>> > REGISTERs to FreeSWITCH boxes), they should accept traffic from OpenSIPS
>> > IP address.
>> > And than you could send calls back to your phones through OpenSIPS too.
>> > "Path" support is needed on FreeSWITCH side, but I hope it works (but I
>> > tested it year ago or more).
>> >
>> > Beste regards,
>> >
>> > kokoska.rokoska
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From pjintheusa at gmail.com  Tue May  4 06:15:22 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Tue, 4 May 2010 09:15:22 -0400
Subject: [Freeswitch-users] Getting the IP (domain) of the FS box a
	phone registered with
In-Reply-To: <i2z59ad9ca11005040604kf8ea18dckd99a842fbfc0966c@mail.gmail.com>
References: <g2i367751821005031753u16511e8ez684f47c7e7f8cf0f@mail.gmail.com>
	<4BDFBCAF.2020108@epbx.cz>
	<66CEF6A4-2E4D-477E-8BE5-37033BC3F1FF@freeswitch.org>
	<w2s367751821005040538sdfabcbf4wc9b1f5340d98dea7@mail.gmail.com>
	<i2z59ad9ca11005040604kf8ea18dckd99a842fbfc0966c@mail.gmail.com>
Message-ID: <o2v367751821005040615z3537eec3s8896fa8db263074e@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Mod_sofia#Specifying_SIP_Proxy_With_fs_path

On Tue, May 4, 2010 at 9:04 AM, Henry Huang <red.rain.seven at gmail.com>wrote:

> Brian:
>
> Where can I find information regarding FreeSWITCH path?
>
> Thanks,
>
> Henry
>
>
> On Tue, May 4, 2010 at 8:38 PM, Phillip Jones <pjintheusa at gmail.com>wrote:
>
>> kokoska,
>>
>> thanks for that. I am considering setting up opensips to deal with mobile
>> phone connections (as well as carrier connections) but it just introduces a
>> single point of failure. I am trying to ing to avoid this.
>>
>>
>> At this point I am wondering whether there is a simple way of finding out
>> which FS box a client is registered with. (without doing a SQL query into
>> the shared DB - although that would be ok too)
>>
>> I am wondering then whether I can use FS_PATH to route through that box
>> (domain)
>>
>> Thanks!
>>
>> Pj
>>
>>
>>
>> On Tue, May 4, 2010 at 2:51 AM, Brian West <brian at freeswitch.org> wrote:
>>
>>> We already have path support in FreeSWITCH.
>>>
>>> /b
>>>
>>>
>>> On May 4, 2010, at 1:20 AM, Martin Dvorak <martin at epbx.cz> wrote:
>>>
>>> > Dne 4.5.2010 2:53, Phillip Jones napsal(a):
>>> >> Hi there,
>>> >>
>>> >> I have a cluster of FS boxes.  OpenSIPS sits in front of the boxes to
>>> >> load balance.
>>> >>
>>> >> The internal profile is sharing the same DB via ODBC.
>>> >>
>>> >> A mobile phone SIP client is registering with FS1, and a call for that
>>> >> SIP client arrives on FS2. The mobile phone will not accept
>>> unsolicited
>>> >> IP traffic, so FS1 must send the invite.
>>> >>
>>> >> From FS2, how do I find the IP (domain) the mobile phone is registered
>>> >> to? Without iterating through all my domains with sofia_contact? Or
>>> >> doing a look up in the DB.
>>> >>
>>> >>
>>> >> Is there a simpler way?
>>> >>
>>> >
>>> > I think yes :-)
>>> >
>>> > If your phones registering "through" OpenSIPS (OpenSIPS just forwards
>>> > REGISTERs to FreeSWITCH boxes), they should accept traffic from
>>> OpenSIPS
>>> > IP address.
>>> > And than you could send calls back to your phones through OpenSIPS too.
>>> > "Path" support is needed on FreeSWITCH side, but I hope it works (but I
>>> > tested it year ago or more).
>>> >
>>> > Beste regards,
>>> >
>>> > kokoska.rokoska
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Henry Huang
> UniC Solution - Communication Unified
> VoIP & Open Source software Consultant
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kenfulmer at icstechnologysolutions.com  Tue May  4 06:29:19 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Tue, 4 May 2010 08:29:19 -0500
Subject: [Freeswitch-users] Different codecs for different devices?
Message-ID: <009601caeb8d$cd5953e0$680bfba0$@com>

This may be a far out question but here goes:

 

Our fax machines that are connected to ATA's need to be g.711 on both Call
Leg A and Call Leg B (in either direction). 

 

However, our phones need to be g.711 on the internal side that faces our
PBX. On the side facing our SIP provider, the calls should be transcoded to
g.729. Inbound the calls to phones (not ATA's) need to be g.729 on the
inbound call leg and g.711 on the outbound call leg. 

 

I realize we can set inbound / outbound codec preferences in single or
multiple profiles and we can set the "absolute_codec_string" value for the
outbound call leg. 

 

However, I've yet to determine how to set the inbound call leg's codec
differently depending on the device. Is this possible? 

 

Thanks,

 

Ken Fulmer

 

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From red.rain.seven at gmail.com  Tue May  4 06:50:56 2010
From: red.rain.seven at gmail.com (Henry Huang)
Date: Tue, 4 May 2010 21:50:56 +0800
Subject: [Freeswitch-users] Getting the IP (domain) of the FS box a
	phone registered with
In-Reply-To: <o2v367751821005040615z3537eec3s8896fa8db263074e@mail.gmail.com>
References: <g2i367751821005031753u16511e8ez684f47c7e7f8cf0f@mail.gmail.com>
	<4BDFBCAF.2020108@epbx.cz>
	<66CEF6A4-2E4D-477E-8BE5-37033BC3F1FF@freeswitch.org>
	<w2s367751821005040538sdfabcbf4wc9b1f5340d98dea7@mail.gmail.com>
	<i2z59ad9ca11005040604kf8ea18dckd99a842fbfc0966c@mail.gmail.com>
	<o2v367751821005040615z3537eec3s8896fa8db263074e@mail.gmail.com>
Message-ID: <m2p59ad9ca11005040650rbb5bd03gd854fc15047c5330@mail.gmail.com>

Phillip:

Thanks. Does the proxy need to be configured in some way to be used as a
path? For instance, if the proxy being used as path is another FS. Does the
FS need to do anything special in the dialplan ? or it's a sofia level
thing?

Henry


On Tue, May 4, 2010 at 9:15 PM, Phillip Jones <pjintheusa at gmail.com> wrote:

> http://wiki.freeswitch.org/wiki/Mod_sofia#Specifying_SIP_Proxy_With_fs_path
>
>
> On Tue, May 4, 2010 at 9:04 AM, Henry Huang <red.rain.seven at gmail.com>wrote:
>
>> Brian:
>>
>> Where can I find information regarding FreeSWITCH path?
>>
>> Thanks,
>>
>> Henry
>>
>>
>> On Tue, May 4, 2010 at 8:38 PM, Phillip Jones <pjintheusa at gmail.com>wrote:
>>
>>> kokoska,
>>>
>>> thanks for that. I am considering setting up opensips to deal with mobile
>>> phone connections (as well as carrier connections) but it just introduces a
>>> single point of failure. I am trying to ing to avoid this.
>>>
>>>
>>> At this point I am wondering whether there is a simple way of finding out
>>> which FS box a client is registered with. (without doing a SQL query into
>>> the shared DB - although that would be ok too)
>>>
>>> I am wondering then whether I can use FS_PATH to route through that box
>>> (domain)
>>>
>>> Thanks!
>>>
>>> Pj
>>>
>>>
>>>
>>> On Tue, May 4, 2010 at 2:51 AM, Brian West <brian at freeswitch.org> wrote:
>>>
>>>> We already have path support in FreeSWITCH.
>>>>
>>>> /b
>>>>
>>>>
>>>> On May 4, 2010, at 1:20 AM, Martin Dvorak <martin at epbx.cz> wrote:
>>>>
>>>> > Dne 4.5.2010 2:53, Phillip Jones napsal(a):
>>>> >> Hi there,
>>>> >>
>>>> >> I have a cluster of FS boxes.  OpenSIPS sits in front of the boxes to
>>>> >> load balance.
>>>> >>
>>>> >> The internal profile is sharing the same DB via ODBC.
>>>> >>
>>>> >> A mobile phone SIP client is registering with FS1, and a call for
>>>> that
>>>> >> SIP client arrives on FS2. The mobile phone will not accept
>>>> unsolicited
>>>> >> IP traffic, so FS1 must send the invite.
>>>> >>
>>>> >> From FS2, how do I find the IP (domain) the mobile phone is
>>>> registered
>>>> >> to? Without iterating through all my domains with sofia_contact? Or
>>>> >> doing a look up in the DB.
>>>> >>
>>>> >>
>>>> >> Is there a simpler way?
>>>> >>
>>>> >
>>>> > I think yes :-)
>>>> >
>>>> > If your phones registering "through" OpenSIPS (OpenSIPS just forwards
>>>> > REGISTERs to FreeSWITCH boxes), they should accept traffic from
>>>> OpenSIPS
>>>> > IP address.
>>>> > And than you could send calls back to your phones through OpenSIPS
>>>> too.
>>>> > "Path" support is needed on FreeSWITCH side, but I hope it works (but
>>>> I
>>>> > tested it year ago or more).
>>>> >
>>>> > Beste regards,
>>>> >
>>>> > kokoska.rokoska
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Henry Huang
>> UniC Solution - Communication Unified
>> VoIP & Open Source software Consultant
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From pjintheusa at gmail.com  Tue May  4 07:24:32 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Tue, 4 May 2010 10:24:32 -0400
Subject: [Freeswitch-users] Getting the IP (domain) of the FS box a
	phone registered with
In-Reply-To: <m2p59ad9ca11005040650rbb5bd03gd854fc15047c5330@mail.gmail.com>
References: <g2i367751821005031753u16511e8ez684f47c7e7f8cf0f@mail.gmail.com>
	<4BDFBCAF.2020108@epbx.cz>
	<66CEF6A4-2E4D-477E-8BE5-37033BC3F1FF@freeswitch.org>
	<w2s367751821005040538sdfabcbf4wc9b1f5340d98dea7@mail.gmail.com>
	<i2z59ad9ca11005040604kf8ea18dckd99a842fbfc0966c@mail.gmail.com>
	<o2v367751821005040615z3537eec3s8896fa8db263074e@mail.gmail.com>
	<m2p59ad9ca11005040650rbb5bd03gd854fc15047c5330@mail.gmail.com>
Message-ID: <o2g367751821005040724p5fc0d0acyed885ac8cf9816bb@mail.gmail.com>

Henry - I don't know - but I will be trying this this afternoon so will let
you know.

Now if I could only find the FS box that a phone is registered with - I
would be golden!



On Tue, May 4, 2010 at 9:50 AM, Henry Huang <red.rain.seven at gmail.com>wrote:

> Phillip:
>
> Thanks. Does the proxy need to be configured in some way to be used as a
> path? For instance, if the proxy being used as path is another FS. Does the
> FS need to do anything special in the dialplan ? or it's a sofia level
> thing?
>
> Henry
>
>
> On Tue, May 4, 2010 at 9:15 PM, Phillip Jones <pjintheusa at gmail.com>wrote:
>
>>
>> http://wiki.freeswitch.org/wiki/Mod_sofia#Specifying_SIP_Proxy_With_fs_path
>>
>>
>> On Tue, May 4, 2010 at 9:04 AM, Henry Huang <red.rain.seven at gmail.com>wrote:
>>
>>> Brian:
>>>
>>> Where can I find information regarding FreeSWITCH path?
>>>
>>> Thanks,
>>>
>>> Henry
>>>
>>>
>>> On Tue, May 4, 2010 at 8:38 PM, Phillip Jones <pjintheusa at gmail.com>wrote:
>>>
>>>> kokoska,
>>>>
>>>> thanks for that. I am considering setting up opensips to deal with
>>>> mobile phone connections (as well as carrier connections) but it just
>>>> introduces a single point of failure. I am trying to ing to avoid this.
>>>>
>>>>
>>>> At this point I am wondering whether there is a simple way of finding
>>>> out which FS box a client is registered with. (without doing a SQL query
>>>> into the shared DB - although that would be ok too)
>>>>
>>>> I am wondering then whether I can use FS_PATH to route through that box
>>>> (domain)
>>>>
>>>> Thanks!
>>>>
>>>> Pj
>>>>
>>>>
>>>>
>>>> On Tue, May 4, 2010 at 2:51 AM, Brian West <brian at freeswitch.org>wrote:
>>>>
>>>>> We already have path support in FreeSWITCH.
>>>>>
>>>>> /b
>>>>>
>>>>>
>>>>> On May 4, 2010, at 1:20 AM, Martin Dvorak <martin at epbx.cz> wrote:
>>>>>
>>>>> > Dne 4.5.2010 2:53, Phillip Jones napsal(a):
>>>>> >> Hi there,
>>>>> >>
>>>>> >> I have a cluster of FS boxes.  OpenSIPS sits in front of the boxes
>>>>> to
>>>>> >> load balance.
>>>>> >>
>>>>> >> The internal profile is sharing the same DB via ODBC.
>>>>> >>
>>>>> >> A mobile phone SIP client is registering with FS1, and a call for
>>>>> that
>>>>> >> SIP client arrives on FS2. The mobile phone will not accept
>>>>> unsolicited
>>>>> >> IP traffic, so FS1 must send the invite.
>>>>> >>
>>>>> >> From FS2, how do I find the IP (domain) the mobile phone is
>>>>> registered
>>>>> >> to? Without iterating through all my domains with sofia_contact? Or
>>>>> >> doing a look up in the DB.
>>>>> >>
>>>>> >>
>>>>> >> Is there a simpler way?
>>>>> >>
>>>>> >
>>>>> > I think yes :-)
>>>>> >
>>>>> > If your phones registering "through" OpenSIPS (OpenSIPS just forwards
>>>>> > REGISTERs to FreeSWITCH boxes), they should accept traffic from
>>>>> OpenSIPS
>>>>> > IP address.
>>>>> > And than you could send calls back to your phones through OpenSIPS
>>>>> too.
>>>>> > "Path" support is needed on FreeSWITCH side, but I hope it works (but
>>>>> I
>>>>> > tested it year ago or more).
>>>>> >
>>>>> > Beste regards,
>>>>> >
>>>>> > kokoska.rokoska
>>>>> >
>>>>> > _______________________________________________
>>>>> > FreeSWITCH-users mailing list
>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> > UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> > http://www.freeswitch.org
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Henry Huang
>>> UniC Solution - Communication Unified
>>> VoIP & Open Source software Consultant
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Henry Huang
> UniC Solution - Communication Unified
> VoIP & Open Source software Consultant
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From douga at cachecomm.com  Tue May  4 09:13:34 2010
From: douga at cachecomm.com (Doug Albrechtsen)
Date: Tue, 04 May 2010 10:13:34 -0600
Subject: [Freeswitch-users] Caller ID Name Display on Phone
In-Reply-To: <AANLkTimR1Z9U_M9csP08YRz94Iv-6S8wT71xB1PBVUUH@mail.gmail.com>
References: <4BDF427B.8000308@cachecomm.com>
	<AANLkTimR1Z9U_M9csP08YRz94Iv-6S8wT71xB1PBVUUH@mail.gmail.com>
Message-ID: <4BE047AE.1010101@cachecomm.com>

Apparently this is an unusual driver configuration we are 
using...otherwise someone certainly would have dealt with it previously.
We are using this config as directed by Sangoma to use their B601DE 
Hybrid Board...and to get the 2B Channel Transfer Feature..otherwise we 
would be more mainstream on this....sorry.

I have opened openzap_libpri.c..doesn't seem to be the right spot to be 
looking.

Could you give me a little more direction on how to go about 
troubleshooting and resolving this problem?

Thanks for your help...FS is awesome!

Anthony Minessale wrote:
> the libpri module for openzap may not be getting the info from the 
> correct field?
> you would have to have a look in the code.
>
>
> On Mon, May 3, 2010 at 4:39 PM, Doug Albrechtsen <douga at cachecomm.com 
> <mailto:douga at cachecomm.com>> wrote:
>
>     We are using FS 1.0 (svn 17188) with Libpri/dahdi/openzap on
>     Sangoma T-1
>     card.
>
>     When a call comes in on the PRI the Calling Number is shown twice
>     on the
>     phone.  Both where the number should be and also in place of the
>     Calling
>     Party Name. Call completes and talks just fine.
>
>     A PRI trace on the FS box shows that the name is being received
>     from the
>     provider in the facility message.
>     Running the "Info" command in the dialplan shows both caller_id_number
>     and caller_id_name containing the Callers phone number.
>
>     Should Caller-ID Name and Number be shown by default or is there a
>     setting that needs to be made to show the Name on the phone?
>
>     Thanks
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From ijurado at econcept.es  Tue May  4 00:11:47 2010
From: ijurado at econcept.es (Isaac Jurado)
Date: Tue, 4 May 2010 09:11:47 +0200
Subject: [Freeswitch-users] Encoding issue with mod_xml_cdr (apparently)
Message-ID: <20100504071146.GA27482@econcept04.econcept.es>

Hi,

We are setting up a web service for mod_xml_cdr but we have problems
with non-ascii characters encoded in UTF-8.  The POST request
(url-encoded) stops its data when it finds the first non-ascii, like the
following:

-----8<-----
<?xml version="1.0"?>
<cdr>
  <channel_data>
    <state>CS_REPORTING</state>
    <direction>inbound</direction>
    <state_number>11</state_number>
    <flags>0=1;18=1;36=1;38=1;51=1</flags>
    <caps>1=1;2=1;3=1</caps>
  </channel_data>
  <variables>
    <uuid>faa6c7e9-9b4a-4ac0-8ebb-971b2e2f8eaa</uuid>
    <sip_local_network_addr>192.168.1.99</sip_local_network_addr>
    <sip_network_ip>192.168.1.99</sip_network_ip>
    <sip_network_port>1024</sip_network_port>
    <sip_received_ip>192.168.1.99</sip_received_ip>
    <sip_received_port>1024</sip_received_port>
    <sip_via_protocol>udp</sip_via_protocol>
    <sip_authorized>true</sip_authorized>
    <sip_number_alias>33</sip_number_alias>
    <sip_auth_username>33</sip_auth_username>
    <sip_auth_realm>econcept.es</sip_auth_realm>
    <number_alias>33</number_alias>
    <user_name>33</user_name>
    <domain_name>econcept.es</domain_name>
    <accountcode>33</accountcode>
    <user_context>econcept.es</user_context>
    <caller_id_name>Peque%%
----->8-----

This is what the web service finds under the 'cdr' POST key.  The last
two percent characters seem to precede the spanish '?'.  The
caller_id_name is supplied at register time by issuing an UTF-8 encoded
XML to feed mod_xml_curl.

I've tried disabling the url-encoding like so (xml_cdr.conf.xml):

    <param name="encode" value="false"/>

But then, the web service does not appear to be receiving anything.

Any ideas?

Cheers.

-- 
Isaac Jurado
Internet Busines Solutions eConcept
http://www.econcept.es


From mark at mdsh.com  Tue May  4 05:00:26 2010
From: mark at mdsh.com (Mark Himsley)
Date: Tue, 04 May 2010 13:00:26 +0100
Subject: [Freeswitch-users] alpha-numeric password
Message-ID: <4BE00C5A.6090201@mdsh.com>

Hi,

I'm very new to Freeswitch - I want to move from Asterisk to Freeswitch
for my home VoIP switch.

In Asterisk sip.conf I have defined extensions like this

[201]
type=friend
username=201
secret=mypass
<...SNIP...>

If I create ${FREESWITCH_CONF}/directory/default/201.xml starting like this:

<include>
  <user id="201">
    <params>
      <param name="password" value="mypass"/>
<...SNIP...>

and just change the server the phone connects to to be my new freeswitch
server then the phone cannot authentcate:

2010-05-04 11:48:28.936358 [WARNING] sofia_reg.c:1030 SIP auth failure
(REGISTER) on sofia profile 'internal' for [201 at 10.0.0.10] from ip
10.0.0.228

But if I set the value element of the password param to a number and
also change the password in the phone to that number then the phone will
authenticate.

Does freeswitch only allow numeric passwords for SIP phones? I can't see
a definition on the freeswitch web site for what is valid for a
password, and all my Googling has failed too :-(

I'd _like_ to keep the passwords unchanged, but I can live with changing
them if needed. I was just wondering if I missed something on the
freeswitch web site.

Thanks in advance.

-- 
Mark


From jerry.richards at teotech.com  Tue May  4 09:25:57 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 4 May 2010 09:25:57 -0700
Subject: [Freeswitch-users] Event Sockets For Log Levels
Message-ID: <6332C520ECE14A2B94288D5CB7BBB1A5@greyhawk.tonecommander.com>

Is there a way to setup an event socket to generate notifications upon log
events (i.e. what normally goes to the FS CLI)?  If so, can it be filtered
by log level?

Thanks,
Jerry



From kenfulmer at icstechnologysolutions.com  Tue May  4 09:32:47 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Tue, 4 May 2010 11:32:47 -0500
Subject: [Freeswitch-users] Different codecs for different devices?
In-Reply-To: <009601caeb8d$cd5953e0$680bfba0$@com>
References: <009601caeb8d$cd5953e0$680bfba0$@com>
Message-ID: <010f01caeba7$6ea4a470$4beded50$@com>

Just in case others are interested, this is possible with the following
parameters:

 

SIP Profile:

 

    <param name="inbound-late-negotiation" value="true"/>

 

Dial-Plan:

 

      <action application="set" data="inherit_codec=true"/>

 

The codec negotiated in Call Leg B, is forced onto Call Leg A. This is
possible due to the "inbound late negotiation" parameter in the sip profile.

 

Hope this helps someone else. 

 

Ken

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken
Fulmer
Sent: Tuesday, May 04, 2010 8:29 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Different codecs for different devices?

 

This may be a far out question but here goes:

 

Our fax machines that are connected to ATA's need to be g.711 on both Call
Leg A and Call Leg B (in either direction). 

 

However, our phones need to be g.711 on the internal side that faces our
PBX. On the side facing our SIP provider, the calls should be transcoded to
g.729. Inbound the calls to phones (not ATA's) need to be g.729 on the
inbound call leg and g.711 on the outbound call leg. 

 

I realize we can set inbound / outbound codec preferences in single or
multiple profiles and we can set the "absolute_codec_string" value for the
outbound call leg. 

 

However, I've yet to determine how to set the inbound call leg's codec
differently depending on the device. Is this possible? 

 

Thanks,

 

Ken Fulmer

 

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From anthony.minessale at gmail.com  Tue May  4 09:45:21 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 4 May 2010 11:45:21 -0500
Subject: [Freeswitch-users] Caller ID Name Display on Phone
In-Reply-To: <4BE047AE.1010101@cachecomm.com>
References: <4BDF427B.8000308@cachecomm.com>
	<AANLkTimR1Z9U_M9csP08YRz94Iv-6S8wT71xB1PBVUUH@mail.gmail.com>
	<4BE047AE.1010101@cachecomm.com>
Message-ID: <AANLkTinAWpa0vKdwoi6yHDs3Tndhbeizbx4mtqv5e-0K@mail.gmail.com>

Since you are using FS you may want to use the sangoma supported PRI stack
that comes with the driver as described on their wiki, then they will
probably be even more willing to help you since it's their code you would be
using and they support it very well.



On Tue, May 4, 2010 at 11:13 AM, Doug Albrechtsen <douga at cachecomm.com>wrote:

> Apparently this is an unusual driver configuration we are
> using...otherwise someone certainly would have dealt with it previously.
> We are using this config as directed by Sangoma to use their B601DE
> Hybrid Board...and to get the 2B Channel Transfer Feature..otherwise we
> would be more mainstream on this....sorry.
>
> I have opened openzap_libpri.c..doesn't seem to be the right spot to be
> looking.
>
> Could you give me a little more direction on how to go about
> troubleshooting and resolving this problem?
>
> Thanks for your help...FS is awesome!
>
> Anthony Minessale wrote:
> > the libpri module for openzap may not be getting the info from the
> > correct field?
> > you would have to have a look in the code.
> >
> >
> > On Mon, May 3, 2010 at 4:39 PM, Doug Albrechtsen <douga at cachecomm.com
> > <mailto:douga at cachecomm.com>> wrote:
> >
> >     We are using FS 1.0 (svn 17188) with Libpri/dahdi/openzap on
> >     Sangoma T-1
> >     card.
> >
> >     When a call comes in on the PRI the Calling Number is shown twice
> >     on the
> >     phone.  Both where the number should be and also in place of the
> >     Calling
> >     Party Name. Call completes and talks just fine.
> >
> >     A PRI trace on the FS box shows that the name is being received
> >     from the
> >     provider in the facility message.
> >     Running the "Info" command in the dialplan shows both
> caller_id_number
> >     and caller_id_name containing the Callers phone number.
> >
> >     Should Caller-ID Name and Number be shown by default or is there a
> >     setting that needs to be made to show the Name on the phone?
> >
> >     Thanks
> >
> >
> >     _______________________________________________
> >     FreeSWITCH-users mailing list
> >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     http://www.freeswitch.org
> >
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> > IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> > pstn:+19193869900
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
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
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From anthony.minessale at gmail.com  Tue May  4 09:48:22 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 4 May 2010 11:48:22 -0500
Subject: [Freeswitch-users] alpha-numeric password
In-Reply-To: <4BE00C5A.6090201@mdsh.com>
References: <4BE00C5A.6090201@mdsh.com>
Message-ID: <AANLkTimWn0SSDF8gbwtu4CL6MSYaQ8fQ61p4ixUONm9v@mail.gmail.com>

There is no such limitation
did you do reloadxml and or restart FS after you changed the configuration?


On Tue, May 4, 2010 at 7:00 AM, Mark Himsley <mark at mdsh.com> wrote:

> Hi,
>
> I'm very new to Freeswitch - I want to move from Asterisk to Freeswitch
> for my home VoIP switch.
>
> In Asterisk sip.conf I have defined extensions like this
>
> [201]
> type=friend
> username=201
> secret=mypass
> <...SNIP...>
>
> If I create ${FREESWITCH_CONF}/directory/default/201.xml starting like
> this:
>
> <include>
>  <user id="201">
>    <params>
>      <param name="password" value="mypass"/>
> <...SNIP...>
>
> and just change the server the phone connects to to be my new freeswitch
> server then the phone cannot authentcate:
>
> 2010-05-04 11:48:28.936358 [WARNING] sofia_reg.c:1030 SIP auth failure
> (REGISTER) on sofia profile 'internal' for [201 at 10.0.0.10] from ip
> 10.0.0.228
>
> But if I set the value element of the password param to a number and
> also change the password in the phone to that number then the phone will
> authenticate.
>
> Does freeswitch only allow numeric passwords for SIP phones? I can't see
> a definition on the freeswitch web site for what is valid for a
> password, and all my Googling has failed too :-(
>
> I'd _like_ to keep the passwords unchanged, but I can live with changing
> them if needed. I was just wondering if I missed something on the
> freeswitch web site.
>
> Thanks in advance.
>
> --
> Mark
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <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
-------------- next part --------------
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From anthony.minessale at gmail.com  Tue May  4 09:50:07 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 4 May 2010 11:50:07 -0500
Subject: [Freeswitch-users] Event Sockets For Log Levels
In-Reply-To: <6332C520ECE14A2B94288D5CB7BBB1A5@greyhawk.tonecommander.com>
References: <6332C520ECE14A2B94288D5CB7BBB1A5@greyhawk.tonecommander.com>
Message-ID: <AANLkTimjRUxCGtjsmUMapHgZQDG_PN4sGlQt9iJatsdz@mail.gmail.com>

yes

log [level_name_or_number]

eg

log debug\n\n


On Tue, May 4, 2010 at 11:25 AM, Jerry Richards
<jerry.richards at teotech.com>wrote:

> Is there a way to setup an event socket to generate notifications upon log
> events (i.e. what normally goes to the FS CLI)?  If so, can it be filtered
> by log level?
>
> Thanks,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <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
-------------- next part --------------
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From anthony.minessale at gmail.com  Tue May  4 09:52:24 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 4 May 2010 11:52:24 -0500
Subject: [Freeswitch-users] Different codecs for different devices?
In-Reply-To: <010f01caeba7$6ea4a470$4beded50$@com>
References: <009601caeb8d$cd5953e0$680bfba0$@com>
	<010f01caeba7$6ea4a470$4beded50$@com>
Message-ID: <AANLkTiluKFA2nqpDr4lUKujb2YFw5vAawYQRjlf7JOc6@mail.gmail.com>

you can regex the SDP for something specific and use that to set the
absolute_codec_string in a standalone extension that comes first in the
stack and has continue=true on it



On Tue, May 4, 2010 at 11:32 AM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  Just in case others are interested, this is possible with the following
> parameters:
>
>
>
> SIP Profile:
>
>
>
>     <param name="inbound-late-negotiation" value="true"/>
>
>
>
> Dial-Plan:
>
>
>
>       <action application="set" data="inherit_codec=true"/>
>
>
>
> The codec negotiated in Call Leg B, is forced onto Call Leg A. This is
> possible due to the ?inbound late negotiation? parameter in the sip profile.
>
>
>
> Hope this helps someone else.
>
>
>
> Ken
>
>
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ken Fulmer
> *Sent:* Tuesday, May 04, 2010 8:29 AM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] Different codecs for different devices?
>
>
>
> This may be a far out question but here goes:
>
>
>
> Our fax machines that are connected to ATA?s need to be g.711 on both Call
> Leg A and Call Leg B (in either direction).
>
>
>
> However, our phones need to be g.711 on the internal side that faces our
> PBX. On the side facing our SIP provider, the calls should be transcoded to
> g.729. Inbound the calls to phones (not ATA?s) need to be g.729 on the
> inbound call leg and g.711 on the outbound call leg.
>
>
>
> I realize we can set inbound / outbound codec preferences in single or
> multiple profiles and we can set the ?absolute_codec_string? value for the
> outbound call leg.
>
>
>
> However, I?ve yet to determine how to set the inbound call leg?s codec
> differently depending on the device. Is this possible?
>
>
>
> Thanks,
>
>
>
> Ken Fulmer
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <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
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From anthony.minessale at gmail.com  Tue May  4 10:12:33 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 4 May 2010 12:12:33 -0500
Subject: [Freeswitch-users] mod_com_g729 DECODER CREATE FAILED
In-Reply-To: <4BDABDA9.5020707@gmx.net>
References: <4BD85C5F.3000302@gmx.net>
	<y2u191c3a031004281324j26173b32nc89a6b0f9716871d@mail.gmail.com>
	<4BDABDA9.5020707@gmx.net>
Message-ID: <AANLkTilHq4lZ_qt9-XjAVbvn7Dlur9u-7a28MERi_Bao@mail.gmail.com>

Peter, every time you email me directly, when I reply it bounces, I have
sent you a few emails that you do not seem to receive.

Essentially if you need to record a g729 call it will cost an additional
license because the recording subsystem needs to force a decoding path so it
can mux the data together etc.

This is what it said:

Hi. This is the qmail-send program at mx0.gmx.net.
I'm afraid I wasn't able to deliver your message to the following addresses.
This is a permanent error; I've given up. Sorry it didn't work out.

<psteinbach at de.lhsgroup.com>:
Sorry,_I_couldn't_find_any_host_named_de.lhsgroup.com._(#5.1.2)/




On Fri, Apr 30, 2010 at 6:23 AM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> I just updated, with the same result:
>
>
> Anthony Minessale schrieb:
> > do you have lastest git HEAD ?
> > can you update and try again?
> >
> >
>
> Here's the log:
>
> 2010-04-30 13:17:21.819392 [DEBUG] sofia.c:4185 Remote SDP:
> v=0
> o=root 929923105 929923106 IN IP4 192.168.178.125
> s=call
> c=IN IP4 192.168.178.125
> t=0 0
> m=audio 60566 RTP/AVP 18 8 0 99 3 101
> a=rtpmap:18 g729/8000
> a=rtpmap:8 pcma/8000
> a=rtpmap:0 pcmu/8000
> a=rtpmap:99 g726-32/8000
> a=rtpmap:3 gsm/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
>
> 2010-04-30 13:17:21.819392 [DEBUG] sofia.c:4174 Channel
> sofia/internal/sip:211 at 192.168.178.125:2048 entering state [ready][200]
> 2010-04-30 13:17:21.819392 [DEBUG] sofia_glue.c:3662 Audio Codec Compare
> [g729:18:8000:20]/[G729:18:8000:20]
> 2010-04-30 13:17:21.819392 [DEBUG] sofia_glue.c:2372 Set Codec
> sofia/internal/sip:211 at 192.168.178.125:2048 G729/8000 20 ms 160 samples
> 2010-04-30 13:17:21.819392 [DEBUG] sofia_glue.c:3595 Set 2833 dtmf send
> payload to 101
> 2010-04-30 13:17:21.819392 [DEBUG] sofia_glue.c:2612 AUDIO RTP
> [sofia/internal/sip:211 at 192.168.178.125:2048] 192.168.178.220 port 12046
> -> 192.168.178.125 port 60566 codec: 18 ms: 20
> 2010-04-30 13:17:21.819392 [DEBUG] switch_rtp.c:1343 Starting timer
> [soft] 160 bytes per 20ms
> 2010-04-30 13:17:21.823983 [DEBUG] sofia_glue.c:2818 Set 2833 dtmf send
> payload to 101
> 2010-04-30 13:17:21.823983 [DEBUG] sofia_glue.c:2823 Set 2833 dtmf
> receive payload to 101
> 2010-04-30 13:17:21.823983 [DEBUG] switch_channel.c:2347 Send signal
> sofia/internal/200 at my.domain [BREAK]
> 2010-04-30 13:17:21.823983 [NOTICE] sofia.c:4754 Channel
> [sofia/internal/sip:211 at 192.168.178.125:2048] has been answered
> 2010-04-30 13:17:21.830046 [DEBUG] sofia_glue.c:2612 AUDIO RTP
> [sofia/internal/200 at my.domain] 192.168.178.220 port 12006 ->
> 192.168.178.50 port 12770 codec: 18 ms: 20
> 2010-04-30 13:17:21.830046 [DEBUG] switch_rtp.c:1343 Starting timer
> [soft] 160 bytes per 20ms
> 2010-04-30 13:17:21.832815 [DEBUG] sofia_glue.c:2818 Set 2833 dtmf send
> payload to 101
> 2010-04-30 13:17:21.832815 [DEBUG] sofia_glue.c:2823 Set 2833 dtmf
> receive payload to 101
> 2010-04-30 13:17:21.832815 [INFO] switch_rtp.c:1173 Activating Secure
> RTP SEND
> 2010-04-30 13:17:21.832815 [DEBUG] switch_core_sqldb.c:1110 Secure Type:
> srtp:AES_CM_128_HMAC_SHA1_32
> 2010-04-30 13:17:21.832815 [INFO] switch_rtp.c:1153 Activating Secure
> RTP RECV
> 2010-04-30 13:17:21.832815 [DEBUG] switch_core_sqldb.c:1110 Secure Type:
> srtp:AES_CM_128_HMAC_SHA1_32
> 2010-04-30 13:17:21.832815 [DEBUG] mod_sofia.c:663 Local SDP
> sofia/internal/200 at my.domain:
> v=0
> o=FreeSWITCH 1272614235 1272614236 IN IP4 192.168.178.220
> s=FreeSWITCH
> c=IN IP4 192.168.178.220
> t=0 0
> m=audio 12006 RTP/SAVP 18 101
> a=rtpmap:18 g729/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> a=crypto:1 AES_CM_128_HMAC_SHA1_32
> inline:OoR/ZSNzik9jFHdcbzXyGXGSO5E3mGT6tvcpvsqK
>
> 2010-04-30 13:17:21.834022 [DEBUG] switch_core_session.c:703 Send signal
> sofia/internal/sip:211 at 192.168.178.125:2048 [BREAK]
> 2010-04-30 13:17:21.834022 [DEBUG] sofia.c:4174 Channel
> sofia/internal/200 at my.domain entering state [completed][200]
> 2010-04-30 13:17:21.834022 [DEBUG] switch_core_session.c:642 Send signal
> sofia/internal/200 at my.domain [BREAK]
> 2010-04-30 13:17:21.835065 [NOTICE] switch_ivr_originate.c:3174 Channel
> [sofia/internal/200 at my.domain] has been answered
> 2010-04-30 13:17:21.835065 [DEBUG] switch_ivr_originate.c:3219 Originate
> Resulted in Success: [sofia/internal/sip:211 at 192.168.178.125:2048]
> 2010-04-30 13:17:21.838531 [DEBUG] switch_ivr_originate.c:3219 Originate
> Resulted in Success: [sofia/internal/sip:211 at 192.168.178.125:2048]
> 2010-04-30 13:17:21.838531 [DEBUG] switch_core_session.c:642 Send signal
> sofia/internal/sip:211 at 192.168.178.125:2048 [BREAK]
> 2010-04-30 13:17:21.839601 [DEBUG] switch_core_session.c:642 Send signal
> sofia/internal/200 at my.domain [BREAK]
> 2010-04-30 13:17:21.839601 [DEBUG] switch_ivr_bridge.c:1182
> (sofia/internal/sip:211 at 192.168.178.125:2048) State Change
> CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
> 2010-04-30 13:17:21.839601 [DEBUG] switch_core_session.c:1022 Send
> signal sofia/internal/sip:211 at 192.168.178.125:2048 [BREAK]
> 2010-04-30 13:17:21.839601 [DEBUG] switch_core_state_machine.c:314
> (sofia/internal/sip:211 at 192.168.178.125:2048) Running State Change
> CS_EXCHANGE_MEDIA
> 2010-04-30 13:17:21.839601 [DEBUG] switch_core_state_machine.c:351
> (sofia/internal/sip:211 at 192.168.178.125:2048) State EXCHANGE_MEDIA
> 2010-04-30 13:17:21.839601 [DEBUG] mod_sofia.c:534 SOFIA EXCHANGE_MEDIA
> 2010-04-30 13:17:21.861445 [DEBUG] switch_core_media_bug.c:360 Attaching
> BUG to sofia/internal/sip:211 at 192.168.178.125:2048
> 2010-04-30 13:17:21.891110 [DEBUG] switch_rtp.c:2443 Correct ip/port
> confirmed.
> 2010-04-30 13:17:21.951915 [DEBUG] switch_rtp.c:2443 Correct ip/port
> confirmed.
> 2010-04-30 13:17:21.951915 [INFO] mod_com_g729.c:146 DECODER CREATE -
> 0x904e070 0x8fb39c0
> 2010-04-30 13:17:22.031586 [INFO] mod_com_g729.c:146 DECODER CREATE -
> 0x8fcd7e8 0x8e448a0
> 2010-04-30 13:17:22.247792 [DEBUG] switch_core_media_bug.c:360 Attaching
> BUG to sofia/internal/200 at my.domain
> 2010-04-30 13:17:22.253759 [ERR] mod_com_g729.c:142 DECODER CREATE
> FAILED - 0x90990a0 (nil)
> 2010-04-30 13:17:22.253759 [ERR] switch_core_io.c:327 Codec G.729
> decoder error!
> 2010-04-30 13:17:22.253759 [DEBUG] switch_ivr_bridge.c:478
> sofia/internal/200 at my.domain ending bridge by request from read function
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <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
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From douga at cachecomm.com  Tue May  4 11:04:12 2010
From: douga at cachecomm.com (Doug Albrechtsen)
Date: Tue, 04 May 2010 12:04:12 -0600
Subject: [Freeswitch-users] Caller ID Name Display on Phone
In-Reply-To: <AANLkTinAWpa0vKdwoi6yHDs3Tndhbeizbx4mtqv5e-0K@mail.gmail.com>
References: <4BDF427B.8000308@cachecomm.com>	<AANLkTimR1Z9U_M9csP08YRz94Iv-6S8wT71xB1PBVUUH@mail.gmail.com>	<4BE047AE.1010101@cachecomm.com>
	<AANLkTinAWpa0vKdwoi6yHDs3Tndhbeizbx4mtqv5e-0K@mail.gmail.com>
Message-ID: <4BE0619C.3030700@cachecomm.com>

Unfortunately, according to Sangoma, we cannot use their PRI stack and 
get the 2B Channel Transfer to work.  It will be supported in a future 
version of their new driver...thus we use this workaround.  Could you 
direct me to a particular source file(s) where this name assignment 
should happen?

Thanks

Anthony Minessale wrote:
>
> Since you are using FS you may want to use the sangoma supported PRI 
> stack that comes with the driver as described on their wiki, then they 
> will probably be even more willing to help you since it's their code 
> you would be using and they support it very well.
>
>
>
> On Tue, May 4, 2010 at 11:13 AM, Doug Albrechtsen <douga at cachecomm.com 
> <mailto:douga at cachecomm.com>> wrote:
>
>     Apparently this is an unusual driver configuration we are
>     using...otherwise someone certainly would have dealt with it
>     previously.
>     We are using this config as directed by Sangoma to use their B601DE
>     Hybrid Board...and to get the 2B Channel Transfer
>     Feature..otherwise we
>     would be more mainstream on this....sorry.
>
>     I have opened openzap_libpri.c..doesn't seem to be the right spot
>     to be
>     looking.
>
>     Could you give me a little more direction on how to go about
>     troubleshooting and resolving this problem?
>
>     Thanks for your help...FS is awesome!
>
>     Anthony Minessale wrote:
>     > the libpri module for openzap may not be getting the info from the
>     > correct field?
>     > you would have to have a look in the code.
>     >
>     >
>     > On Mon, May 3, 2010 at 4:39 PM, Doug Albrechtsen
>     <douga at cachecomm.com <mailto:douga at cachecomm.com>
>     > <mailto:douga at cachecomm.com <mailto:douga at cachecomm.com>>> wrote:
>     >
>     >     We are using FS 1.0 (svn 17188) with Libpri/dahdi/openzap on
>     >     Sangoma T-1
>     >     card.
>     >
>     >     When a call comes in on the PRI the Calling Number is shown
>     twice
>     >     on the
>     >     phone.  Both where the number should be and also in place of the
>     >     Calling
>     >     Party Name. Call completes and talks just fine.
>     >
>     >     A PRI trace on the FS box shows that the name is being received
>     >     from the
>     >     provider in the facility message.
>     >     Running the "Info" command in the dialplan shows both
>     caller_id_number
>     >     and caller_id_name containing the Callers phone number.
>     >
>     >     Should Caller-ID Name and Number be shown by default or is
>     there a
>     >     setting that needs to be made to show the Name on the phone?
>     >
>     >     Thanks
>     >
>     >
>     >     _______________________________________________
>     >     FreeSWITCH-users mailing list
>     >     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>     >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >    
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >     http://www.freeswitch.org
>     >
>     >
>     >
>     >
>     > --
>     > Anthony Minessale II
>     >
>     > FreeSWITCH http://www.freeswitch.org/
>     > ClueCon http://www.cluecon.com/
>     > Twitter: http://twitter.com/FreeSWITCH_wire
>     >
>     > AIM: anthm
>     > MSN:anthony_minessale at hotmail.com
>     <mailto:MSN%3Aanthony_minessale at hotmail.com>
>     > <mailto:MSN%3Aanthony_minessale at hotmail.com
>     <mailto:MSN%253Aanthony_minessale at hotmail.com>>
>     > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>     <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>     > <mailto:PAYPAL%3Aanthony.minessale at gmail.com
>     <mailto:PAYPAL%253Aanthony.minessale at gmail.com>>
>     > IRC: irc.freenode.net <http://irc.freenode.net>
>     <http://irc.freenode.net> #freeswitch
>     >
>     > FreeSWITCH Developer Conference
>     > sip:888 at conference.freeswitch.org
>     <mailto:sip%3A888 at conference.freeswitch.org>
>     > <mailto:sip%3A888 at conference.freeswitch.org
>     <mailto:sip%253A888 at conference.freeswitch.org>>
>     > googletalk:conf+888 at conference.freeswitch.org
>     <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>     > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org
>     <mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org>>
>     > pstn:+19193869900
>     >
>     ------------------------------------------------------------------------
>     >
>     > _______________________________________________
>     > FreeSWITCH-users mailing list
>     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > http://www.freeswitch.org
>     >
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From anthony.minessale at gmail.com  Tue May  4 11:26:34 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 4 May 2010 13:26:34 -0500
Subject: [Freeswitch-users] mod_com_g729 DECODER CREATE FAILED
In-Reply-To: <AANLkTilHq4lZ_qt9-XjAVbvn7Dlur9u-7a28MERi_Bao@mail.gmail.com>
References: <4BD85C5F.3000302@gmx.net>
	<y2u191c3a031004281324j26173b32nc89a6b0f9716871d@mail.gmail.com>
	<4BDABDA9.5020707@gmx.net>
	<AANLkTilHq4lZ_qt9-XjAVbvn7Dlur9u-7a28MERi_Bao@mail.gmail.com>
Message-ID: <AANLkTimPZkIXxd0PIoqxyM9L5KNU7__F2TYiMLmCqK7h@mail.gmail.com>

Did you real my last email?

Do not send me email directly unless its from a valid email address that I
can reply to.
I cannot reply to you on the direct email because it bounces back to me
EVERY TIME.

you need to fix your return email address.


On Tue, May 4, 2010 at 12:12 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Peter, every time you email me directly, when I reply it bounces, I have
> sent you a few emails that you do not seem to receive.
>
> Essentially if you need to record a g729 call it will cost an additional
> license because the recording subsystem needs to force a decoding path so it
> can mux the data together etc.
>
> This is what it said:
>
> Hi. This is the qmail-send program at mx0.gmx.net.
> I'm afraid I wasn't able to deliver your message to the following
> addresses.
> This is a permanent error; I've given up. Sorry it didn't work out.
>
> <psteinbach at de.lhsgroup.com>:
> Sorry,_I_couldn't_find_any_host_named_de.lhsgroup.com._(#5.1.2)/
>
>
>
>
> On Fri, Apr 30, 2010 at 6:23 AM, Peter P GMX <Prometheus001 at gmx.net>wrote:
>
>> I just updated, with the same result:
>>
>>
>> Anthony Minessale schrieb:
>> > do you have lastest git HEAD ?
>> > can you update and try again?
>> >
>> >
>>
>> Here's the log:
>>
>> 2010-04-30 13:17:21.819392 [DEBUG] sofia.c:4185 Remote SDP:
>> v=0
>> o=root 929923105 929923106 IN IP4 192.168.178.125
>> s=call
>> c=IN IP4 192.168.178.125
>> t=0 0
>> m=audio 60566 RTP/AVP 18 8 0 99 3 101
>> a=rtpmap:18 g729/8000
>> a=rtpmap:8 pcma/8000
>> a=rtpmap:0 pcmu/8000
>> a=rtpmap:99 g726-32/8000
>> a=rtpmap:3 gsm/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>>
>> 2010-04-30 13:17:21.819392 [DEBUG] sofia.c:4174 Channel
>> sofia/internal/sip:211 at 192.168.178.125:2048 entering state [ready][200]
>> 2010-04-30 13:17:21.819392 [DEBUG] sofia_glue.c:3662 Audio Codec Compare
>> [g729:18:8000:20]/[G729:18:8000:20]
>> 2010-04-30 13:17:21.819392 [DEBUG] sofia_glue.c:2372 Set Codec
>> sofia/internal/sip:211 at 192.168.178.125:2048 G729/8000 20 ms 160 samples
>> 2010-04-30 13:17:21.819392 [DEBUG] sofia_glue.c:3595 Set 2833 dtmf send
>> payload to 101
>> 2010-04-30 13:17:21.819392 [DEBUG] sofia_glue.c:2612 AUDIO RTP
>> [sofia/internal/sip:211 at 192.168.178.125:2048] 192.168.178.220 port 12046
>> -> 192.168.178.125 port 60566 codec: 18 ms: 20
>> 2010-04-30 13:17:21.819392 [DEBUG] switch_rtp.c:1343 Starting timer
>> [soft] 160 bytes per 20ms
>> 2010-04-30 13:17:21.823983 [DEBUG] sofia_glue.c:2818 Set 2833 dtmf send
>> payload to 101
>> 2010-04-30 13:17:21.823983 [DEBUG] sofia_glue.c:2823 Set 2833 dtmf
>> receive payload to 101
>> 2010-04-30 13:17:21.823983 [DEBUG] switch_channel.c:2347 Send signal
>> sofia/internal/200 at my.domain [BREAK]
>> 2010-04-30 13:17:21.823983 [NOTICE] sofia.c:4754 Channel
>> [sofia/internal/sip:211 at 192.168.178.125:2048] has been answered
>> 2010-04-30 13:17:21.830046 [DEBUG] sofia_glue.c:2612 AUDIO RTP
>> [sofia/internal/200 at my.domain] 192.168.178.220 port 12006 ->
>> 192.168.178.50 port 12770 codec: 18 ms: 20
>> 2010-04-30 13:17:21.830046 [DEBUG] switch_rtp.c:1343 Starting timer
>> [soft] 160 bytes per 20ms
>> 2010-04-30 13:17:21.832815 [DEBUG] sofia_glue.c:2818 Set 2833 dtmf send
>> payload to 101
>> 2010-04-30 13:17:21.832815 [DEBUG] sofia_glue.c:2823 Set 2833 dtmf
>> receive payload to 101
>> 2010-04-30 13:17:21.832815 [INFO] switch_rtp.c:1173 Activating Secure
>> RTP SEND
>> 2010-04-30 13:17:21.832815 [DEBUG] switch_core_sqldb.c:1110 Secure Type:
>> srtp:AES_CM_128_HMAC_SHA1_32
>> 2010-04-30 13:17:21.832815 [INFO] switch_rtp.c:1153 Activating Secure
>> RTP RECV
>> 2010-04-30 13:17:21.832815 [DEBUG] switch_core_sqldb.c:1110 Secure Type:
>> srtp:AES_CM_128_HMAC_SHA1_32
>> 2010-04-30 13:17:21.832815 [DEBUG] mod_sofia.c:663 Local SDP
>> sofia/internal/200 at my.domain:
>> v=0
>> o=FreeSWITCH 1272614235 1272614236 IN IP4 192.168.178.220
>> s=FreeSWITCH
>> c=IN IP4 192.168.178.220
>> t=0 0
>> m=audio 12006 RTP/SAVP 18 101
>> a=rtpmap:18 g729/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=silenceSupp:off - - - -
>> a=ptime:20
>> a=sendrecv
>> a=crypto:1 AES_CM_128_HMAC_SHA1_32
>> inline:OoR/ZSNzik9jFHdcbzXyGXGSO5E3mGT6tvcpvsqK
>>
>> 2010-04-30 13:17:21.834022 [DEBUG] switch_core_session.c:703 Send signal
>> sofia/internal/sip:211 at 192.168.178.125:2048 [BREAK]
>> 2010-04-30 13:17:21.834022 [DEBUG] sofia.c:4174 Channel
>> sofia/internal/200 at my.domain entering state [completed][200]
>> 2010-04-30 13:17:21.834022 [DEBUG] switch_core_session.c:642 Send signal
>> sofia/internal/200 at my.domain [BREAK]
>> 2010-04-30 13:17:21.835065 [NOTICE] switch_ivr_originate.c:3174 Channel
>> [sofia/internal/200 at my.domain] has been answered
>> 2010-04-30 13:17:21.835065 [DEBUG] switch_ivr_originate.c:3219 Originate
>> Resulted in Success: [sofia/internal/sip:211 at 192.168.178.125:2048]
>> 2010-04-30 13:17:21.838531 [DEBUG] switch_ivr_originate.c:3219 Originate
>> Resulted in Success: [sofia/internal/sip:211 at 192.168.178.125:2048]
>> 2010-04-30 13:17:21.838531 [DEBUG] switch_core_session.c:642 Send signal
>> sofia/internal/sip:211 at 192.168.178.125:2048 [BREAK]
>> 2010-04-30 13:17:21.839601 [DEBUG] switch_core_session.c:642 Send signal
>> sofia/internal/200 at my.domain [BREAK]
>> 2010-04-30 13:17:21.839601 [DEBUG] switch_ivr_bridge.c:1182
>> (sofia/internal/sip:211 at 192.168.178.125:2048) State Change
>> CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA
>> 2010-04-30 13:17:21.839601 [DEBUG] switch_core_session.c:1022 Send
>> signal sofia/internal/sip:211 at 192.168.178.125:2048 [BREAK]
>> 2010-04-30 13:17:21.839601 [DEBUG] switch_core_state_machine.c:314
>> (sofia/internal/sip:211 at 192.168.178.125:2048) Running State Change
>> CS_EXCHANGE_MEDIA
>> 2010-04-30 13:17:21.839601 [DEBUG] switch_core_state_machine.c:351
>> (sofia/internal/sip:211 at 192.168.178.125:2048) State EXCHANGE_MEDIA
>> 2010-04-30 13:17:21.839601 [DEBUG] mod_sofia.c:534 SOFIA EXCHANGE_MEDIA
>> 2010-04-30 13:17:21.861445 [DEBUG] switch_core_media_bug.c:360 Attaching
>> BUG to sofia/internal/sip:211 at 192.168.178.125:2048
>> 2010-04-30 13:17:21.891110 [DEBUG] switch_rtp.c:2443 Correct ip/port
>> confirmed.
>> 2010-04-30 13:17:21.951915 [DEBUG] switch_rtp.c:2443 Correct ip/port
>> confirmed.
>> 2010-04-30 13:17:21.951915 [INFO] mod_com_g729.c:146 DECODER CREATE -
>> 0x904e070 0x8fb39c0
>> 2010-04-30 13:17:22.031586 [INFO] mod_com_g729.c:146 DECODER CREATE -
>> 0x8fcd7e8 0x8e448a0
>> 2010-04-30 13:17:22.247792 [DEBUG] switch_core_media_bug.c:360 Attaching
>> BUG to sofia/internal/200 at my.domain
>> 2010-04-30 13:17:22.253759 [ERR] mod_com_g729.c:142 DECODER CREATE
>> FAILED - 0x90990a0 (nil)
>> 2010-04-30 13:17:22.253759 [ERR] switch_core_io.c:327 Codec G.729
>> decoder error!
>> 2010-04-30 13:17:22.253759 [DEBUG] switch_ivr_bridge.c:478
>> sofia/internal/200 at my.domain ending bridge by request from read function
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <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
>



-- 
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
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From anthony.minessale at gmail.com  Tue May  4 11:29:11 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 4 May 2010 13:29:11 -0500
Subject: [Freeswitch-users] Caller ID Name Display on Phone
In-Reply-To: <4BE0619C.3030700@cachecomm.com>
References: <4BDF427B.8000308@cachecomm.com>
	<AANLkTimR1Z9U_M9csP08YRz94Iv-6S8wT71xB1PBVUUH@mail.gmail.com>
	<4BE047AE.1010101@cachecomm.com>
	<AANLkTinAWpa0vKdwoi6yHDs3Tndhbeizbx4mtqv5e-0K@mail.gmail.com>
	<4BE0619C.3030700@cachecomm.com>
Message-ID: <AANLkTim7WcRP2nRT8hc2iCYwgm1H1sy0kDVxzITrgrTy@mail.gmail.com>

libs/openzap/src/ozmod/ozmod_libpri/ozmod_libpri.c line 776


On Tue, May 4, 2010 at 1:04 PM, Doug Albrechtsen <douga at cachecomm.com>wrote:

> Unfortunately, according to Sangoma, we cannot use their PRI stack and
> get the 2B Channel Transfer to work.  It will be supported in a future
> version of their new driver...thus we use this workaround.  Could you
> direct me to a particular source file(s) where this name assignment
> should happen?
>
> Thanks
>
> Anthony Minessale wrote:
> >
> > Since you are using FS you may want to use the sangoma supported PRI
> > stack that comes with the driver as described on their wiki, then they
> > will probably be even more willing to help you since it's their code
> > you would be using and they support it very well.
> >
> >
> >
> > On Tue, May 4, 2010 at 11:13 AM, Doug Albrechtsen <douga at cachecomm.com
> > <mailto:douga at cachecomm.com>> wrote:
> >
> >     Apparently this is an unusual driver configuration we are
> >     using...otherwise someone certainly would have dealt with it
> >     previously.
> >     We are using this config as directed by Sangoma to use their B601DE
> >     Hybrid Board...and to get the 2B Channel Transfer
> >     Feature..otherwise we
> >     would be more mainstream on this....sorry.
> >
> >     I have opened openzap_libpri.c..doesn't seem to be the right spot
> >     to be
> >     looking.
> >
> >     Could you give me a little more direction on how to go about
> >     troubleshooting and resolving this problem?
> >
> >     Thanks for your help...FS is awesome!
> >
> >     Anthony Minessale wrote:
> >     > the libpri module for openzap may not be getting the info from the
> >     > correct field?
> >     > you would have to have a look in the code.
> >     >
> >     >
> >     > On Mon, May 3, 2010 at 4:39 PM, Doug Albrechtsen
> >     <douga at cachecomm.com <mailto:douga at cachecomm.com>
> >     > <mailto:douga at cachecomm.com <mailto:douga at cachecomm.com>>> wrote:
> >     >
> >     >     We are using FS 1.0 (svn 17188) with Libpri/dahdi/openzap on
> >     >     Sangoma T-1
> >     >     card.
> >     >
> >     >     When a call comes in on the PRI the Calling Number is shown
> >     twice
> >     >     on the
> >     >     phone.  Both where the number should be and also in place of
> the
> >     >     Calling
> >     >     Party Name. Call completes and talks just fine.
> >     >
> >     >     A PRI trace on the FS box shows that the name is being received
> >     >     from the
> >     >     provider in the facility message.
> >     >     Running the "Info" command in the dialplan shows both
> >     caller_id_number
> >     >     and caller_id_name containing the Callers phone number.
> >     >
> >     >     Should Caller-ID Name and Number be shown by default or is
> >     there a
> >     >     setting that needs to be made to show the Name on the phone?
> >     >
> >     >     Thanks
> >     >
> >     >
> >     >     _______________________________________________
> >     >     FreeSWITCH-users mailing list
> >     >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
> >     >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     >
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     >     http://www.freeswitch.org
> >     >
> >     >
> >     >
> >     >
> >     > --
> >     > Anthony Minessale II
> >     >
> >     > FreeSWITCH http://www.freeswitch.org/
> >     > ClueCon http://www.cluecon.com/
> >     > Twitter: http://twitter.com/FreeSWITCH_wire
> >     >
> >     > AIM: anthm
> >     > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >     <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> >     > <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >     <mailto:MSN%253Aanthony_minessale at hotmail.com<MSN%25253Aanthony_minessale at hotmail.com>
> >>
> >     > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >     <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> >     > <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >     <mailto:PAYPAL%253Aanthony.minessale at gmail.com<PAYPAL%25253Aanthony.minessale at gmail.com>
> >>
> >     > IRC: irc.freenode.net <http://irc.freenode.net>
> >     <http://irc.freenode.net> #freeswitch
> >     >
> >     > FreeSWITCH Developer Conference
> >     > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >     <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> >     > <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >     <mailto:sip%253A888 at conference.freeswitch.org<sip%25253A888 at conference.freeswitch.org>
> >>
> >     > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >     <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> >     > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >     <mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org<googletalk%25253Aconf%25252B888 at conference.freeswitch.org>
> >>
> >     > pstn:+19193869900
> >     >
> >
> ------------------------------------------------------------------------
> >     >
> >     > _______________________________________________
> >     > FreeSWITCH-users mailing list
> >     > FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     >
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     > http://www.freeswitch.org
> >     >
> >
> >
> >     _______________________________________________
> >     FreeSWITCH-users mailing list
> >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     http://www.freeswitch.org
> >
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> > IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> > pstn:+19193869900
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
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
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From steve.d.ward at gmail.com  Tue May  4 11:55:44 2010
From: steve.d.ward at gmail.com (Steven Ward)
Date: Tue, 4 May 2010 14:55:44 -0400
Subject: [Freeswitch-users] Method to force a transfer of A-Leg
In-Reply-To: <EC5C7D5E-DC08-4A5E-B93B-F9D42EF9EE5E@jerris.com>
References: <p2w4ea6e8f21004210709w54a18d7ao6bb9adb2acaaca8a@mail.gmail.com>
	<EC5C7D5E-DC08-4A5E-B93B-F9D42EF9EE5E@jerris.com>
Message-ID: <w2m4ea6e8f21005041155x6a4ccbbgc64cdbb59114c790@mail.gmail.com>

Sorry I haven't provided feedback any sooner...

I've been looking at uuid_transfer and uuid_deflect, and I'm still having a
tough time getting the desired effect for my situation.

I hope these details make things a bit more clear.

The thing is, this script is monitoring transfers that are done by a
Polycom; the transfer starts as attended, but then is converted to a blind
xfer (so the xfer is ringing, and I hit the Transfer button again to exit
out of the transfer).  So the Polycom sends a REFER to the a-leg w/ a
REPLACES param.

My script then turns on, sleeps for a bit, and then checks out the a-leg.
If the a-leg hasn't been answered, I'd like the script to cancel that
transfer and move the leg somewhere else on the box.

The thing is, the a-leg isn't bridged yet because it hasn't been answered
(destination does not even send early media).  And the a-leg doesn't even
know in its channel variables what channel it's trying to ring, because of
the way the transfer works.  The a-leg is just waiting for the transfer to
complete.  As Tony graciously described to me on irc one time, until the
destination answers (or sends early media), the channel that dialed out to
do the transfer remains up in a zombified state.  And once the destination
answers, then the zombified channel drops out and the a-leg is bridged to
the destination.

Incidentally, I do see that FS keeps track of the "zombie" channel in the
a-leg's *att_xfer_kill_uuid* channel variable.

The impact of this to what I'm trying to do is that uuid_transfer doesn't
work for taking over the call.  I can uuid_transfer the a-leg, but the
original destination of the transfer keeps ringing and the
att_xfer_kill_uuid channel stays up.  So now it's as if the a-leg is ringing
two destinations, and the original destination never stops ringing.

Is there any way I can find out what channel the a-leg is trying to get
connected to via att_xfer_kill_uuid?  Or is there any way I can truly cancel
that attended xfer and take control of the a-leg?

I see that uuid_deflect can be used to send a REFER and send it off-box, but
I'd like to simply send the a-leg through dialplan on-box.  (And when I've
tried using uuid_deflect to REFER it back to another destination on my same
box, I *sometimes* get a busy for a reason I haven't been able to debug
yet.)

So, any thoughts on controlling the Polycom attended-transfer-turned-blind
:) would be much appreciated.

This is for an application that replicates the features of an operator
console that is currently in use - that's why I need to control calls so
maniacally.  Once I get this polished off, I look forward to putting the
final results onto the wiki as an example application.

Thanks again for all the help I've already received on irc and the list in
the past, and thanks for any comments on this scenario...

- Steve
(sward_)
steve.d.ward at gmail.com


On Sat, Apr 24, 2010 at 2:47 PM, Michael Jerris <mike at jerris.com> wrote:

> just uuid_transfer the a leg.
>
> On Apr 21, 2010, at 10:09 AM, Steven Ward wrote:
>
> Hello all,
>
> I have a lua script running that checks the state of a call between A and B
> - the call between A and B was set up through a Polycom's attended transfer
> feature, so A itself didn't necessarily execute the bridge to B.
>
> A gets early media from B.  After a certain amount of time, if A is still
> getting early media, I want the script to end that call between A and B, and
> send A through some specific dialplan.
>
> How does uuid_transfer work for this goal?  I'd like to kill B and move A
> to specific dialplan.  I don't want B to be in the transfer at all.
>
> I know there may be many possibilities here; I'm just wondering if anyone
> can recommend something.  Thanks.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Tue May  4 12:40:45 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 4 May 2010 14:40:45 -0500
Subject: [Freeswitch-users] Method to force a transfer of A-Leg
In-Reply-To: <w2m4ea6e8f21005041155x6a4ccbbgc64cdbb59114c790@mail.gmail.com>
References: <p2w4ea6e8f21004210709w54a18d7ao6bb9adb2acaaca8a@mail.gmail.com>
	<EC5C7D5E-DC08-4A5E-B93B-F9D42EF9EE5E@jerris.com>
	<w2m4ea6e8f21005041155x6a4ccbbgc64cdbb59114c790@mail.gmail.com>
Message-ID: <AANLkTilzLGTOxy7aKWW669j3h-_ayis5cu6PPWOli3fY@mail.gmail.com>

Could you just collect the digits from the user in the ivr, transfer him
back to the conference.
then originate the call separately with the collected digits to enter the
conference once it answers?


On Tue, May 4, 2010 at 1:55 PM, Steven Ward <steve.d.ward at gmail.com> wrote:

> Sorry I haven't provided feedback any sooner...
>
> I've been looking at uuid_transfer and uuid_deflect, and I'm still having a
> tough time getting the desired effect for my situation.
>
> I hope these details make things a bit more clear.
>
> The thing is, this script is monitoring transfers that are done by a
> Polycom; the transfer starts as attended, but then is converted to a blind
> xfer (so the xfer is ringing, and I hit the Transfer button again to exit
> out of the transfer).  So the Polycom sends a REFER to the a-leg w/ a
> REPLACES param.
>
> My script then turns on, sleeps for a bit, and then checks out the a-leg.
> If the a-leg hasn't been answered, I'd like the script to cancel that
> transfer and move the leg somewhere else on the box.
>
> The thing is, the a-leg isn't bridged yet because it hasn't been answered
> (destination does not even send early media).  And the a-leg doesn't even
> know in its channel variables what channel it's trying to ring, because of
> the way the transfer works.  The a-leg is just waiting for the transfer to
> complete.  As Tony graciously described to me on irc one time, until the
> destination answers (or sends early media), the channel that dialed out to
> do the transfer remains up in a zombified state.  And once the destination
> answers, then the zombified channel drops out and the a-leg is bridged to
> the destination.
>
> Incidentally, I do see that FS keeps track of the "zombie" channel in the
> a-leg's *att_xfer_kill_uuid* channel variable.
>
> The impact of this to what I'm trying to do is that uuid_transfer doesn't
> work for taking over the call.  I can uuid_transfer the a-leg, but the
> original destination of the transfer keeps ringing and the
> att_xfer_kill_uuid channel stays up.  So now it's as if the a-leg is ringing
> two destinations, and the original destination never stops ringing.
>
> Is there any way I can find out what channel the a-leg is trying to get
> connected to via att_xfer_kill_uuid?  Or is there any way I can truly cancel
> that attended xfer and take control of the a-leg?
>
> I see that uuid_deflect can be used to send a REFER and send it off-box,
> but I'd like to simply send the a-leg through dialplan on-box.  (And when
> I've tried using uuid_deflect to REFER it back to another destination on my
> same box, I *sometimes* get a busy for a reason I haven't been able to debug
> yet.)
>
> So, any thoughts on controlling the Polycom attended-transfer-turned-blind
> :) would be much appreciated.
>
> This is for an application that replicates the features of an operator
> console that is currently in use - that's why I need to control calls so
> maniacally.  Once I get this polished off, I look forward to putting the
> final results onto the wiki as an example application.
>
> Thanks again for all the help I've already received on irc and the list in
> the past, and thanks for any comments on this scenario...
>
> - Steve
> (sward_)
> steve.d.ward at gmail.com
>
>
> On Sat, Apr 24, 2010 at 2:47 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> just uuid_transfer the a leg.
>>
>> On Apr 21, 2010, at 10:09 AM, Steven Ward wrote:
>>
>> Hello all,
>>
>> I have a lua script running that checks the state of a call between A and
>> B - the call between A and B was set up through a Polycom's attended
>> transfer feature, so A itself didn't necessarily execute the bridge to B.
>>
>> A gets early media from B.  After a certain amount of time, if A is still
>> getting early media, I want the script to end that call between A and B, and
>> send A through some specific dialplan.
>>
>> How does uuid_transfer work for this goal?  I'd like to kill B and move A
>> to specific dialplan.  I don't want B to be in the transfer at all.
>>
>> I know there may be many possibilities here; I'm just wondering if anyone
>> can recommend something.  Thanks.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <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
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From steve.d.ward at gmail.com  Tue May  4 12:53:18 2010
From: steve.d.ward at gmail.com (Steven Ward)
Date: Tue, 4 May 2010 15:53:18 -0400
Subject: [Freeswitch-users] Method to force a transfer of A-Leg
In-Reply-To: <AANLkTilzLGTOxy7aKWW669j3h-_ayis5cu6PPWOli3fY@mail.gmail.com>
References: <p2w4ea6e8f21004210709w54a18d7ao6bb9adb2acaaca8a@mail.gmail.com>
	<EC5C7D5E-DC08-4A5E-B93B-F9D42EF9EE5E@jerris.com>
	<w2m4ea6e8f21005041155x6a4ccbbgc64cdbb59114c790@mail.gmail.com>
	<AANLkTilzLGTOxy7aKWW669j3h-_ayis5cu6PPWOli3fY@mail.gmail.com>
Message-ID: <m2p4ea6e8f21005041253y8dcd7190pb3e2c00c49ce4ecf@mail.gmail.com>

At the moment, I'm not using an ivr.  The transfer is done by the phone (so
when I hit the Transfer button, the phone puts a-leg on hold and I dial a
new call; I hit the Transfer button again and the Polycom does the REFER w/
REPLACES and hangs up on a-leg and the new destination).

(Makes me wish the phone didn't come with its own built-in Transfer feature
with accompanying button and I could have FS handle everything... :-)

On Tue, May 4, 2010 at 3:40 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Could you just collect the digits from the user in the ivr, transfer him
> back to the conference.
> then originate the call separately with the collected digits to enter the
> conference once it answers?
>
>
> On Tue, May 4, 2010 at 1:55 PM, Steven Ward <steve.d.ward at gmail.com>wrote:
>
>> Sorry I haven't provided feedback any sooner...
>>
>> I've been looking at uuid_transfer and uuid_deflect, and I'm still having
>> a tough time getting the desired effect for my situation.
>>
>> I hope these details make things a bit more clear.
>>
>> The thing is, this script is monitoring transfers that are done by a
>> Polycom; the transfer starts as attended, but then is converted to a blind
>> xfer (so the xfer is ringing, and I hit the Transfer button again to exit
>> out of the transfer).  So the Polycom sends a REFER to the a-leg w/ a
>> REPLACES param.
>>
>> My script then turns on, sleeps for a bit, and then checks out the a-leg.
>> If the a-leg hasn't been answered, I'd like the script to cancel that
>> transfer and move the leg somewhere else on the box.
>>
>> The thing is, the a-leg isn't bridged yet because it hasn't been answered
>> (destination does not even send early media).  And the a-leg doesn't even
>> know in its channel variables what channel it's trying to ring, because of
>> the way the transfer works.  The a-leg is just waiting for the transfer to
>> complete.  As Tony graciously described to me on irc one time, until the
>> destination answers (or sends early media), the channel that dialed out to
>> do the transfer remains up in a zombified state.  And once the destination
>> answers, then the zombified channel drops out and the a-leg is bridged to
>> the destination.
>>
>> Incidentally, I do see that FS keeps track of the "zombie" channel in the
>> a-leg's *att_xfer_kill_uuid* channel variable.
>>
>> The impact of this to what I'm trying to do is that uuid_transfer doesn't
>> work for taking over the call.  I can uuid_transfer the a-leg, but the
>> original destination of the transfer keeps ringing and the
>> att_xfer_kill_uuid channel stays up.  So now it's as if the a-leg is ringing
>> two destinations, and the original destination never stops ringing.
>>
>> Is there any way I can find out what channel the a-leg is trying to get
>> connected to via att_xfer_kill_uuid?  Or is there any way I can truly cancel
>> that attended xfer and take control of the a-leg?
>>
>> I see that uuid_deflect can be used to send a REFER and send it off-box,
>> but I'd like to simply send the a-leg through dialplan on-box.  (And when
>> I've tried using uuid_deflect to REFER it back to another destination on my
>> same box, I *sometimes* get a busy for a reason I haven't been able to debug
>> yet.)
>>
>> So, any thoughts on controlling the Polycom attended-transfer-turned-blind
>> :) would be much appreciated.
>>
>> This is for an application that replicates the features of an operator
>> console that is currently in use - that's why I need to control calls so
>> maniacally.  Once I get this polished off, I look forward to putting the
>> final results onto the wiki as an example application.
>>
>> Thanks again for all the help I've already received on irc and the list in
>> the past, and thanks for any comments on this scenario...
>>
>> - Steve
>> (sward_)
>> steve.d.ward at gmail.com
>>
>>
>> On Sat, Apr 24, 2010 at 2:47 PM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> just uuid_transfer the a leg.
>>>
>>> On Apr 21, 2010, at 10:09 AM, Steven Ward wrote:
>>>
>>> Hello all,
>>>
>>> I have a lua script running that checks the state of a call between A and
>>> B - the call between A and B was set up through a Polycom's attended
>>> transfer feature, so A itself didn't necessarily execute the bridge to B.
>>>
>>> A gets early media from B.  After a certain amount of time, if A is still
>>> getting early media, I want the script to end that call between A and B, and
>>> send A through some specific dialplan.
>>>
>>> How does uuid_transfer work for this goal?  I'd like to kill B and move A
>>> to specific dialplan.  I don't want B to be in the transfer at all.
>>>
>>> I know there may be many possibilities here; I'm just wondering if anyone
>>> can recommend something.  Thanks.
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue May  4 12:54:16 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 4 May 2010 12:54:16 -0700
Subject: [Freeswitch-users] DTMF stopped working
In-Reply-To: <9C882B8FF4674A898FF079744EC95629@D810>
References: <9C882B8FF4674A898FF079744EC95629@D810>
Message-ID: <s2w87f2f3b91005041254y542193b2le6222c67522cf59b@mail.gmail.com>

I'd recommend getting a pcap of the incoming call and then listen for DTMFs
in the audio stream using Wireshark. If you don't hear the DTMFs in the pcap
then you know that the digits aren't even making it to you and that you'll
need to talk to your carrier. If the DTMFs are present then you will need to
dig a little deeper on your end.

Check this page for some tips:
http://wiki.freeswitch.org/wiki/Packet_Capture

-MC

On Tue, May 4, 2010 at 5:00 AM, Andy <andy at fabulous4.co.uk> wrote:

>  Hi folks,
>
> What would cause DTMF to suddenly stop working on inbound calls? I have a
> relatively simple setup with folks diallg in and navigating through an IVR
> menu. I'm using start_dtmf in the dialplan and I can see this being called
> at the start of the call. Basically everything was working fine before the
> weeked, nothing has changed though we did have some problems with our
> internet connection and now none of the dtmf tones in the incoming calls are
> being indentified by freeswitch. Any clues?
>
> Many thanks
> Andy
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sean at obscuradigital.com  Tue May  4 13:12:45 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Tue, 04 May 2010 13:12:45 -0700
Subject: [Freeswitch-users] Caller id
Message-ID: <C805CDCD.9627%sean@obscuradigital.com>

Ok hate to ask this question because it?s probably  been answered already,
but here goes.....

I?ve got effective caller id and outbound_caller_id param for both name and
number setup in my dialplan
I?ve setup this string in my outgoing bridge dialplan {sip_cid_type=pid} and
{sip_cid_type=rpid}
Also I?ve tried this,
value=?{origination_caller_id_name=?Name',origination_caller_id_number=xxxxx
xxxx}sofia/gateway/trunk_1/$1?

I?ve searched for a solution for this on google and the wiki but none of the
above combinations seem to work.  I?ve only getting the number id but not a
name.

Thanks in advance
Sean
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From msc at freeswitch.org  Tue May  4 13:15:23 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 4 May 2010 13:15:23 -0700
Subject: [Freeswitch-users] Call for assistance: Spanish sound files,
	translations
Message-ID: <i2k87f2f3b91005041315p8375079co4245b4727da0788f@mail.gmail.com>

?Hola mis amigos!

We could use some help with the Spanish sound files that Carlos Reyna has
provided. What we need the most is someone to get the latest phrase_es.xml
file and compare it to the list of sound files that we actually have
recorded. We need a list of sounds that are missing. Also, we need someone
to get the latest phrase_en.xml and compare it to phrase_es.xml. There are
many new English sound prompts that aren't even in the other language phrase
files. We'd like to rectify that.

Additionally, if you are able to translate the phrase_en.xml file into other
languages that would be very welcome. It would be good to get the existing
non-English files updated.

Please contact me off list if you are able to assist with any of the items
listed above.

Thanks,
Michael
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From msc at freeswitch.org  Tue May  4 13:20:06 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 4 May 2010 13:20:06 -0700
Subject: [Freeswitch-users] Caller id
In-Reply-To: <C805CDCD.9627%sean@obscuradigital.com>
References: <C805CDCD.9627%sean@obscuradigital.com>
Message-ID: <i2q87f2f3b91005041320h25f7dacbya5c563c5b40991f1@mail.gmail.com>

On Tue, May 4, 2010 at 1:12 PM, Sean Holt <sean at obscuradigital.com> wrote:

>  Ok hate to ask this question because it?s probably  been answered
> already, but here goes.....
>
> I?ve got effective caller id and outbound_caller_id param for both name and
> number setup in my dialplan
> I?ve setup this string in my outgoing bridge dialplan {sip_cid_type=pid}
> and {sip_cid_type=rpid}
> Also I?ve tried this,
> value=?{origination_caller_id_name=?Name',origination_caller_id_number=xxxxxxxxx}sofia/gateway/trunk_1/$1?
>
> I?ve searched for a solution for this on google and the wiki but none of
> the above combinations seem to work.  I?ve only getting the number id but
> not a name.
>
> origination_caller_id_name/number is for doing origination at the CLI:

originate {origination_caller_id_number=123}sofia/gateway/foo/bar

effective_caller_id_name/number is for a bridge:

<action application="bridge"
data="{effective_caller_id_number=123}sofia/gateway/foo/bar"/>

You can also watch the SIP traffic to see what the INVITEs look like.
-MC
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From sean at obscuradigital.com  Tue May  4 13:49:20 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Tue, 04 May 2010 13:49:20 -0700
Subject: [Freeswitch-users] Caller id
In-Reply-To: <i2q87f2f3b91005041320h25f7dacbya5c563c5b40991f1@mail.gmail.com>
Message-ID: <C805D660.962E%sean@obscuradigital.com>

Ok I made the change but still same issue, no Name in place of the number

Here?s my dialplan entry
 <extension name="main_trunk_1" continue="true">
      <condition field="destination_number" expression="^01{2}([0-9].+)$"
test="global">
        <action application="bridge"
data="{effective_caller_id_name='Name'}sofia/gateway/trunk_1/${prepend}$1"/>
      </condition>
    </extension>

Thanks


On 5/4/10 1:20 PM, "Michael Collins" <msc at freeswitch.org> wrote:

> 
> 
> On Tue, May 4, 2010 at 1:12 PM, Sean Holt <sean at obscuradigital.com> wrote:
>> Ok hate to ask this question because it?s probably ?been answered already,
>> but here goes.....
>> 
>> I?ve got effective caller id and outbound_caller_id param for both name and
>> number setup in my dialplan
>> I?ve setup this string in my outgoing bridge dialplan {sip_cid_type=pid} and
>> {sip_cid_type=rpid}
>> Also I?ve tried this,
>> value=?{origination_caller_id_name=?Name',origination_caller_id_number=xxxxxx
>> xxx}sofia/gateway/trunk_1/$1?
>> 
>> I?ve searched for a solution for this on google and the wiki but none of the
>> above combinations seem to work. ?I?ve only getting the number id but not a
>> name.
>> 
> origination_caller_id_name/number is for doing origination at the CLI:
> 
> originate {origination_caller_id_number=123}sofia/gateway/foo/bar
> 
> effective_caller_id_name/number is for a bridge:
> 
> <action application="bridge"
> data="{effective_caller_id_number=123}sofia/gateway/foo/bar"/>
> 
> You can also watch the SIP traffic to see what the INVITEs look like.
> -MC
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From peder at networkoblivion.com  Tue May  4 14:19:20 2010
From: peder at networkoblivion.com (Peder)
Date: Tue, 4 May 2010 16:19:20 -0500
Subject: [Freeswitch-users] Caller id
In-Reply-To: <C805D660.962E%sean@obscuradigital.com>
References: <i2q87f2f3b91005041320h25f7dacbya5c563c5b40991f1@mail.gmail.com>
	<C805D660.962E%sean@obscuradigital.com>
Message-ID: <149e01caebcf$76371740$62a545c0$@com>

When you say you aren't getting the name, where do you mean?  When you call
someone on the PSTN?  Or on the console debug of FS?  Or on the gateway box?
If you are calling someone on the PSTN, you can't send them names.  You can
only send numbers.

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sean
Holt
Sent: Tuesday, May 04, 2010 3:49 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Caller id

 

Ok I made the change but still same issue, no Name in place of the number

Here's my dialplan entry
 <extension name="main_trunk_1" continue="true">
      <condition field="destination_number" expression="^01{2}([0-9].+)$"
test="global">
        <action application="bridge"
data="{effective_caller_id_name='Name'}sofia/gateway/trunk_1/${prepend}$1"/>
      </condition>
    </extension>

Thanks


On 5/4/10 1:20 PM, "Michael Collins" <msc at freeswitch.org> wrote:



On Tue, May 4, 2010 at 1:12 PM, Sean Holt <sean at obscuradigital.com> wrote:

Ok hate to ask this question because it's probably  been answered already,
but here goes.....

I've got effective caller id and outbound_caller_id param for both name and
number setup in my dialplan
I've setup this string in my outgoing bridge dialplan {sip_cid_type=pid} and
{sip_cid_type=rpid}
Also I've tried this,
value="{origination_caller_id_name='Name',origination_caller_id_number=xxxxx
xxxx}sofia/gateway/trunk_1/$1"

I've searched for a solution for this on google and the wiki but none of the
above combinations seem to work.  I've only getting the number id but not a
name.

origination_caller_id_name/number is for doing origination at the CLI:

originate {origination_caller_id_number=123}sofia/gateway/foo/bar

effective_caller_id_name/number is for a bridge:

<action application="bridge"
data="{effective_caller_id_number=123}sofia/gateway/foo/bar"/>

You can also watch the SIP traffic to see what the INVITEs look like.
-MC



  _____  

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From jmesquita at freeswitch.org  Tue May  4 14:50:46 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Tue, 4 May 2010 18:50:46 -0300
Subject: [Freeswitch-users] Different codecs for different devices?
In-Reply-To: <AANLkTiluKFA2nqpDr4lUKujb2YFw5vAawYQRjlf7JOc6@mail.gmail.com>
References: <009601caeb8d$cd5953e0$680bfba0$@com>
	<010f01caeba7$6ea4a470$4beded50$@com>
	<AANLkTiluKFA2nqpDr4lUKujb2YFw5vAawYQRjlf7JOc6@mail.gmail.com>
Message-ID: <i2xebbc2d551005041450qfec08896l527fa2ee3c8a1639@mail.gmail.com>

I think you can also use another variable on the A leg SDP to make it easier
on the regex. ep_codec_string.

This will contain the parsed version of the A leg SDP and that's what I use
most of the times for these tricks. Also, can you document that? :-)

Regards,
Jo?o Mesquita


On Tue, May 4, 2010 at 1:52 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> you can regex the SDP for something specific and use that to set the
> absolute_codec_string in a standalone extension that comes first in the
> stack and has continue=true on it
>
>
>
> On Tue, May 4, 2010 at 11:32 AM, Ken Fulmer <
> kenfulmer at icstechnologysolutions.com> wrote:
>
>>  Just in case others are interested, this is possible with the following
>> parameters:
>>
>>
>>
>> SIP Profile:
>>
>>
>>
>>     <param name="inbound-late-negotiation" value="true"/>
>>
>>
>>
>> Dial-Plan:
>>
>>
>>
>>       <action application="set" data="inherit_codec=true"/>
>>
>>
>>
>> The codec negotiated in Call Leg B, is forced onto Call Leg A. This is
>> possible due to the ?inbound late negotiation? parameter in the sip profile.
>>
>>
>>
>> Hope this helps someone else.
>>
>>
>>
>> Ken
>>
>>
>>
>>
>>
>>
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ken Fulmer
>> *Sent:* Tuesday, May 04, 2010 8:29 AM
>>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Subject:* [Freeswitch-users] Different codecs for different devices?
>>
>>
>>
>> This may be a far out question but here goes:
>>
>>
>>
>> Our fax machines that are connected to ATA?s need to be g.711 on both Call
>> Leg A and Call Leg B (in either direction).
>>
>>
>>
>> However, our phones need to be g.711 on the internal side that faces our
>> PBX. On the side facing our SIP provider, the calls should be transcoded to
>> g.729. Inbound the calls to phones (not ATA?s) need to be g.729 on the
>> inbound call leg and g.711 on the outbound call leg.
>>
>>
>>
>> I realize we can set inbound / outbound codec preferences in single or
>> multiple profiles and we can set the ?absolute_codec_string? value for the
>> outbound call leg.
>>
>>
>>
>> However, I?ve yet to determine how to set the inbound call leg?s codec
>> differently depending on the device. Is this possible?
>>
>>
>>
>> Thanks,
>>
>>
>>
>> Ken Fulmer
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kenfulmer at icstechnologysolutions.com  Tue May  4 15:11:43 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Tue, 4 May 2010 17:11:43 -0500
Subject: [Freeswitch-users] Annex B
Message-ID: <01af01caebd6$c80c04c0$58240e40$@com>

When we send calls to an Adtran gateway, our calls using g.729 work
properly. I noticed in the SDP the following field:

 

a=fmtp:18 annexb=yes

 

When we send calls to a Cisco gateway, the same call fails. 

 

Inbound g.729 calls coming from the Cisco gateway work properly and I
noticed the following:

 

a=fmtp:18 annexb=no

 

Is there a way to set this parameter in the dial-plan or sip profiles so we
can match what the Cisco gateway is looking for? 

 

Thanks,

 

Ken Fulmer

 

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From mike at jerris.com  Tue May  4 15:12:08 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 4 May 2010 18:12:08 -0400
Subject: [Freeswitch-users] mod_xml_cdr error log is not working
In-Reply-To: <i2p7abab2411005031405o7438559bu9bc450aca3f93e35@mail.gmail.com>
References: <g2p7abab2411004291856s2c2d5018g103f33cf2be9dd4b@mail.gmail.com>
	<DD0949EC-0C93-4A8F-9EA4-C6A669AE93FF@jerris.com>
	<l2z7abab2411005030343xdbe1827coc7e531214aa78864@mail.gmail.com>
	<68DF5166-42A6-4316-9F96-3FD3CC6C9B62@jerris.com>
	<v2w7abab2411005031403v5e00a8ccx7f2f2a5dadef3b29@mail.gmail.com>
	<i2p7abab2411005031405o7438559bu9bc450aca3f93e35@mail.gmail.com>
Message-ID: <4C4850C4-3704-4646-B962-675FFF54B363@jerris.com>

so is this actually a bug as well?

Mike

On May 3, 2010, at 5:05 PM, Frank Church wrote:

> PS. Brian has updated the config samples, http://lists.freeswitch.org/pipermail/freeswitch-users/2010-May/057475.html
> 
> On 3 May 2010 22:03, Frank Church <vfclists at googlemail.com> wrote:
> 
> 
> I think it is a bug because the wiki states that
> log-http-and-disk 	 Default behaviour is to write either HTTP or Disk on HTTP failure. Setting this to true will write to both HTTP and Disk regardless (handy for realtime + reconciliation later if required) 	 true
> 
> But it wasn't writing the HTTP connection errors to the disk until I added log-http-and-disk to the configuration. log-http-and-disk is for writing the CDR itself, but logging HTTP errors did not work until it was added
> 
> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-May/057475.html
> 
> On 3 May 2010 21:14, Michael Jerris <mike at jerris.com> wrote:
> would you mind making sure this gets documented on the wiki, and, if it is not in the sample configuration, send us a patch for that via http://jira.freeswitch.org ?
> 
> Mike

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From brian at freeswitch.org  Tue May  4 15:19:22 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 4 May 2010 17:19:22 -0500
Subject: [Freeswitch-users] Annex B
In-Reply-To: <01af01caebd6$c80c04c0$58240e40$@com>
References: <01af01caebd6$c80c04c0$58240e40$@com>
Message-ID: <F831BDC6-59B9-4539-B288-572504D5F605@freeswitch.org>

<action application="export" data="sip_append_audio_sdp=a=fmtp:18 annexb=no"/>


/b

On May 4, 2010, at 5:11 PM, Ken Fulmer wrote:

> When we send calls to an Adtran gateway, our calls using g.729 work properly. I noticed in the SDP the following field:
>  
> a=fmtp:18 annexb=yes
>  
> When we send calls to a Cisco gateway, the same call fails.
>  
> Inbound g.729 calls coming from the Cisco gateway work properly and I noticed the following:
>  
> a=fmtp:18 annexb=no
>  
> Is there a way to set this parameter in the dial-plan or sip profiles so we can match what the Cisco gateway is looking for?
>  
> Thanks,
>  
> Ken Fulmer
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From kenfulmer at icstechnologysolutions.com  Tue May  4 15:38:17 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Tue, 4 May 2010 17:38:17 -0500
Subject: [Freeswitch-users] Annex B
In-Reply-To: <F831BDC6-59B9-4539-B288-572504D5F605@freeswitch.org>
References: <01af01caebd6$c80c04c0$58240e40$@com>
	<F831BDC6-59B9-4539-B288-572504D5F605@freeswitch.org>
Message-ID: <01c001caebda$7d9f7300$78de5900$@com>

Wow, that worked great. Thanks so much...that's a lifesaver for us!

Out of curiosity, where should I look in the documentation for this
parameter? I didn't see it on the channel variable page. I may be looking in
the wrong places and hate to keep asking you guys so many questions. 

Ken


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Tuesday, May 04, 2010 5:19 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Annex B

<action application="export" data="sip_append_audio_sdp=a=fmtp:18
annexb=no"/>


/b

On May 4, 2010, at 5:11 PM, Ken Fulmer wrote:

> When we send calls to an Adtran gateway, our calls using g.729 work
properly. I noticed in the SDP the following field:
>  
> a=fmtp:18 annexb=yes
>  
> When we send calls to a Cisco gateway, the same call fails.
>  
> Inbound g.729 calls coming from the Cisco gateway work properly and I
noticed the following:
>  
> a=fmtp:18 annexb=no
>  
> Is there a way to set this parameter in the dial-plan or sip profiles so
we can match what the Cisco gateway is looking for?
>  
> Thanks,
>  
> Ken Fulmer
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


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



From brian at freeswitch.org  Tue May  4 16:05:29 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 4 May 2010 18:05:29 -0500
Subject: [Freeswitch-users] Annex B
In-Reply-To: <01c001caebda$7d9f7300$78de5900$@com>
References: <01af01caebd6$c80c04c0$58240e40$@com>
	<F831BDC6-59B9-4539-B288-572504D5F605@freeswitch.org>
	<01c001caebda$7d9f7300$78de5900$@com>
Message-ID: <911ACB36-F863-4C76-9467-B6A2FEDFC0E6@freeswitch.org>

I'm highly involved in all the code so most of the time I have the solution in my chat logs or I know where to go look in the code to see it.  I however do not have the time to write in-depth documentation but however will answer questions if someone writes docs.

/b

On May 4, 2010, at 5:38 PM, Ken Fulmer wrote:

> Wow, that worked great. Thanks so much...that's a lifesaver for us!
> 
> Out of curiosity, where should I look in the documentation for this
> parameter? I didn't see it on the channel variable page. I may be looking in
> the wrong places and hate to keep asking you guys so many questions. 
> 
> Ken



From mrene_lists at avgs.ca  Tue May  4 16:16:59 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Tue, 4 May 2010 19:16:59 -0400
Subject: [Freeswitch-users] Annex B
In-Reply-To: <911ACB36-F863-4C76-9467-B6A2FEDFC0E6@freeswitch.org>
References: <01af01caebd6$c80c04c0$58240e40$@com>
	<F831BDC6-59B9-4539-B288-572504D5F605@freeswitch.org>
	<01c001caebda$7d9f7300$78de5900$@com>
	<911ACB36-F863-4C76-9467-B6A2FEDFC0E6@freeswitch.org>
Message-ID: <68A2CF78-F6AB-4D70-8F49-E2EA362F8B99@avgs.ca>

Lets work on a spotlight module to search Brian's chat logs from freeswitch.org

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




On 2010-05-04, at 7:05 PM, Brian West wrote:

> I'm highly involved in all the code so most of the time I have the solution in my chat logs or I know where to go look in the code to see it.  I however do not have the time to write in-depth documentation but however will answer questions if someone writes docs.
> 
> /b
> 
> On May 4, 2010, at 5:38 PM, Ken Fulmer wrote:
> 
>> Wow, that worked great. Thanks so much...that's a lifesaver for us!
>> 
>> Out of curiosity, where should I look in the documentation for this
>> parameter? I didn't see it on the channel variable page. I may be looking in
>> the wrong places and hate to keep asking you guys so many questions. 
>> 
>> Ken
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Tue May  4 16:21:23 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 4 May 2010 18:21:23 -0500
Subject: [Freeswitch-users] Annex B
In-Reply-To: <68A2CF78-F6AB-4D70-8F49-E2EA362F8B99@avgs.ca>
References: <01af01caebd6$c80c04c0$58240e40$@com>
	<F831BDC6-59B9-4539-B288-572504D5F605@freeswitch.org>
	<01c001caebda$7d9f7300$78de5900$@com>
	<911ACB36-F863-4C76-9467-B6A2FEDFC0E6@freeswitch.org>
	<68A2CF78-F6AB-4D70-8F49-E2EA362F8B99@avgs.ca>
Message-ID: <139A3BAE-6616-44BF-8CA2-D8F685C1A767@freeswitch.org>

Not sure you want to search all I have in my logs :P

/b

On May 4, 2010, at 6:16 PM, Mathieu Rene wrote:

> Lets work on a spotlight module to search Brian's chat logs from freeswitch.org
> 
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
> 
> 

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From babak.freeswitch at gmail.com  Tue May  4 22:19:43 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Wed, 5 May 2010 09:49:43 +0430
Subject: [Freeswitch-users] run error after building in vs 2008!
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C5576CE4DA1@cooper>
References: <i2v1fdf92721004302253l7dbdf254p1ff2b3884ec5a4d4@mail.gmail.com>
	<l2o367751821005010612j46281f6fg17629c38bcd07f4@mail.gmail.com>
	<n2i1fdf92721005011131o221b6dd4sf483e00613963d5@mail.gmail.com>
	<x2z367751821005011229g3b6c217kbda766daac32a44d@mail.gmail.com>
	<E141A983-9EC5-4C62-8D17-675DC0A67009@jerris.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5576CE4D97@cooper>
	<k2y87f2f3b91005031727j1661d31vf76dc19d9e3354e5@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5576CE4DA1@cooper>
Message-ID: <n2n1fdf92721005042219x95407e2cp9a0496b95dee3308@mail.gmail.com>

Thanks for the file
and after download where it should be placed??
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From david.ponzone at gmail.com  Tue May  4 22:41:46 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Wed, 5 May 2010 07:41:46 +0200
Subject: [Freeswitch-users] Caller id
In-Reply-To: <C805D660.962E%sean@obscuradigital.com>
References: <C805D660.962E%sean@obscuradigital.com>
Message-ID: <075E8FCD-8AD0-4889-A86E-CEE423086CBF@gmail.com>

Sean

Peder made a point.
Even if you send call through SIP, I guess some ITSPs don't accept the  
CLID name.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 04/05/2010 ? 22:49, Sean Holt a ?crit :

> Ok I made the change but still same issue, no Name in place of the  
> number
>
> Here?s my dialplan entry
>  <extension name="main_trunk_1" continue="true">
>       <condition field="destination_number" expression="^01{2}([0-9]. 
> +)$" test="global">
>         <action application="bridge"  
> data="{effective_caller_id_name='Name'}sofia/gateway/trunk_1/$ 
> {prepend}$1"/>
>       </condition>
>     </extension>
>
> Thanks
>
>
> On 5/4/10 1:20 PM, "Michael Collins" <msc at freeswitch.org> wrote:
>
>>
>>
>> On Tue, May 4, 2010 at 1:12 PM, Sean Holt <sean at obscuradigital.com>  
>> wrote:
>>> Ok hate to ask this question because it?s probably  been answered  
>>> already, but here goes.....
>>>
>>> I?ve got effective caller id and outbound_caller_id param for both  
>>> name and number setup in my dialplan
>>> I?ve setup this string in my outgoing bridge dialplan  
>>> {sip_cid_type=pid} and {sip_cid_type=rpid}
>>> Also I?ve tried this,  
>>> value 
>>> = 
>>> ?{origination_caller_id_name 
>>> =?Name',origination_caller_id_number=xxxxxxxxx}sofia/gateway/ 
>>> trunk_1/$1?
>>>
>>> I?ve searched for a solution for this on google and the wiki but  
>>> none of the above combinations seem to work.  I?ve only getting  
>>> the number id but not a name.
>>>
>> origination_caller_id_name/number is for doing origination at the  
>> CLI:
>>
>> originate {origination_caller_id_number=123}sofia/gateway/foo/bar
>>
>> effective_caller_id_name/number is for a bridge:
>>
>> <action application="bridge"  
>> data="{effective_caller_id_number=123}sofia/gateway/foo/bar"/>
>>
>> You can also watch the SIP traffic to see what the INVITEs look like.
>> -MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From ovvenkatesan at gmail.com  Tue May  4 22:46:31 2010
From: ovvenkatesan at gmail.com (ovvenkat)
Date: Wed, 5 May 2010 11:16:31 +0530
Subject: [Freeswitch-users] run error after building in vs 2008!
In-Reply-To: <n2n1fdf92721005042219x95407e2cp9a0496b95dee3308@mail.gmail.com>
References: <i2v1fdf92721004302253l7dbdf254p1ff2b3884ec5a4d4@mail.gmail.com>
	<l2o367751821005010612j46281f6fg17629c38bcd07f4@mail.gmail.com>
	<n2i1fdf92721005011131o221b6dd4sf483e00613963d5@mail.gmail.com>
	<x2z367751821005011229g3b6c217kbda766daac32a44d@mail.gmail.com>
	<E141A983-9EC5-4C62-8D17-675DC0A67009@jerris.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5576CE4D97@cooper>
	<k2y87f2f3b91005031727j1661d31vf76dc19d9e3354e5@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5576CE4DA1@cooper>
	<n2n1fdf92721005042219x95407e2cp9a0496b95dee3308@mail.gmail.com>
Message-ID: <n2o47d63d921005042246m6e506ab8s403be7f167229611@mail.gmail.com>

Hi to all,

We have developed a IVR Application in freeSwitch SDK. Now , we want to
deploye it on the production server. So, we  I bought Sangoma A101 T1/E1
card. Now we are facing  a new problem like

*FreeSwitch wont support "Europe ISDN" connection*. their technical support
guys are saying that, we need to buy a additional software called  "*
NetBorderExpress*" to overcome this problem. Is this true? why its so?

Regards
Venkatesan OV.
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From brian at freeswitch.org  Tue May  4 22:52:29 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 5 May 2010 00:52:29 -0500
Subject: [Freeswitch-users] run error after building in vs 2008!
In-Reply-To: <n2o47d63d921005042246m6e506ab8s403be7f167229611@mail.gmail.com>
References: <i2v1fdf92721004302253l7dbdf254p1ff2b3884ec5a4d4@mail.gmail.com>
	<l2o367751821005010612j46281f6fg17629c38bcd07f4@mail.gmail.com>
	<n2i1fdf92721005011131o221b6dd4sf483e00613963d5@mail.gmail.com>
	<x2z367751821005011229g3b6c217kbda766daac32a44d@mail.gmail.com>
	<E141A983-9EC5-4C62-8D17-675DC0A67009@jerris.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5576CE4D97@cooper>
	<k2y87f2f3b91005031727j1661d31vf76dc19d9e3354e5@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5576CE4DA1@cooper>
	<n2n1fdf92721005042219x95407e2cp9a0496b95dee3308@mail.gmail.com>
	<n2o47d63d921005042246m6e506ab8s403be7f167229611@mail.gmail.com>
Message-ID: <3C83F15A-B796-4FF8-982B-7C427CC36F5B@freeswitch.org>

That's a bull, I was told today it does support euroisdn and the wiki seems to agree with me.  

/b  

Sent from my iPad

On May 5, 2010, at 12:46 AM, ovvenkat <ovvenkatesan at gmail.com> wrote:

> Hi to all, 
> 
> We have developed a IVR Application in freeSwitch SDK. Now , we want to  deploye it on the production server. So, we  I bought Sangoma A101 T1/E1 card. Now we are facing  a new problem like 
> 
> FreeSwitch wont support "Europe ISDN" connection. their technical support guys are saying that, we need to buy a additional software called  "NetBorderExpress" to overcome this problem. Is this true? why its so?
> 
> Regards
> Venkatesan OV.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From sean at obscuradigital.com  Tue May  4 23:02:22 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Tue, 04 May 2010 23:02:22 -0700
Subject: [Freeswitch-users] Caller id
In-Reply-To: <075E8FCD-8AD0-4889-A86E-CEE423086CBF@gmail.com>
Message-ID: <C80657FE.9689%sean@obscuradigital.com>

Thanks appreciate the help.


On 5/4/10 10:41 PM, "David Ponzone" <david.ponzone at gmail.com> wrote:

> Sean
> 
> Peder made a point.
> Even if you send call through SIP, I guess some ITSPs don't accept the CLID
> name.
> 
>  
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>   -   www.ipeva-studio.com
> <BLOCKED::http://www.ipeva-studio.com/>
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non
> autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a
> ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
>  
> 
> Le 04/05/2010 ? 22:49, Sean Holt a ?crit :
> 
>>  Ok I made the change but still same issue, no Name in place of the number
>>  
>>  Here?s my dialplan entry
>>   <extension name="main_trunk_1" continue="true">
>>        <condition field="destination_number" expression="^01{2}([0-9].+)$"
>> test="global">
>>          <action application="bridge"
>> data="{effective_caller_id_name='Name'}sofia/gateway/trunk_1/${prepend}$1"/>
>>        </condition>
>>      </extension>
>>  
>>  Thanks
>>  
>>  
>>  On 5/4/10 1:20 PM, "Michael Collins" <msc at freeswitch.org> wrote:
>>  
>>  
>>> 
>>>  
>>>  On Tue, May 4, 2010 at 1:12 PM, Sean Holt <sean at obscuradigital.com> wrote:
>>>  
>>>> Ok hate to ask this question because it?s probably  been answered already,
>>>> but here goes.....
>>>>  
>>>>  I?ve got effective caller id and outbound_caller_id param for both name
>>>> and number setup in my dialplan
>>>>  I?ve setup this string in my outgoing bridge dialplan {sip_cid_type=pid}
>>>> and {sip_cid_type=rpid}
>>>>  Also I?ve tried this,
>>>> value=?{origination_caller_id_name=?Name',origination_caller_id_number=xxxx
>>>> xxxxx}sofia/gateway/trunk_1/$1?
>>>>  
>>>>  I?ve searched for a solution for this on google and the wiki but none of
>>>> the above combinations seem to work.  I?ve only getting the number id but
>>>> not a name.
>>>>  
>>>>  
>>> origination_caller_id_name/number is for doing origination at the CLI:
>>>  
>>>  originate {origination_caller_id_number=123}sofia/gateway/foo/bar
>>>  
>>>  effective_caller_id_name/number is for a bridge:
>>>  
>>>  <action application="bridge"
>>> data="{effective_caller_id_number=123}sofia/gateway/foo/bar"/>
>>>  
>>>  You can also watch the SIP traffic to see what the INVITEs look like.
>>>  -MC
>>>  
>>>  
>>>  
>>> 
>>> _______________________________________________
>>>  FreeSWITCH-users mailing list
>>>  FreeSWITCH-users at lists.freeswitch.org
>>>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>  http://www.freeswitch.org
>>>  
>>  
>>   _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From ovvenkatesan at gmail.com  Tue May  4 23:03:37 2010
From: ovvenkatesan at gmail.com (ovvenkat)
Date: Wed, 5 May 2010 11:33:37 +0530
Subject: [Freeswitch-users] problem in SDK while deploying app on the
	production server
Message-ID: <k2y47d63d921005042303oef5a58d9kd70fddf5ea42306d@mail.gmail.com>

Hi Brian,

thanks you very much for your quick reply. Can you send me the wiki link of
the same.


Regards,
Venkat.

On Wed, May 5, 2010 at 11:22 AM, Brian West <brian at freeswitch.org> wrote:

> That's a bull, I was told today it does support euroisdn and the wiki seems
> to agree with me.
>
> /b
>
> Sent from my iPad
>
> On May 5, 2010, at 12:46 AM, ovvenkat <ovvenkatesan at gmail.com> wrote:
>
> Hi to all,
>
> We have developed a IVR Application in freeSwitch SDK. Now , we want to
> deploye it on the production server. So, we  I bought Sangoma A101 T1/E1
> card. Now we are facing  a new problem like
>
> *FreeSwitch wont support "Europe ISDN" connection*. their technical
> support guys are saying that, we need to buy a additional software called  "
> *NetBorderExpress*" to overcome this problem. Is this true? why its so?
>
> Regards
> Venkatesan OV.
>
> _______________________________________________
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
Regards
Venkatesan OV.
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From msc at freeswitch.org  Tue May  4 23:51:49 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 4 May 2010 23:51:49 -0700
Subject: [Freeswitch-users] Annex B
In-Reply-To: <139A3BAE-6616-44BF-8CA2-D8F685C1A767@freeswitch.org>
References: <01af01caebd6$c80c04c0$58240e40$@com>
	<F831BDC6-59B9-4539-B288-572504D5F605@freeswitch.org>
	<01c001caebda$7d9f7300$78de5900$@com>
	<911ACB36-F863-4C76-9467-B6A2FEDFC0E6@freeswitch.org>
	<68A2CF78-F6AB-4D70-8F49-E2EA362F8B99@avgs.ca>
	<139A3BAE-6616-44BF-8CA2-D8F685C1A767@freeswitch.org>
Message-ID: <r2j87f2f3b91005042351rfda2eb02w407283c35e6f8658@mail.gmail.com>

On Tue, May 4, 2010 at 4:21 PM, Brian West <brian at freeswitch.org> wrote:

> Not sure you want to search all I have in my logs :P
>
Hehe, I'm dead sure that I don't. :D

Ken, maybe you could document this on the wiki. We have a codecs page as
well as a g.729 page. Either one might be appropriate.
-MC


>
> /b
>
> On May 4, 2010, at 6:16 PM, Mathieu Rene wrote:
>
> Lets work on a spotlight module to search Brian's chat logs from
> freeswitch.org
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From peter.olsson at visionutveckling.se  Tue May  4 23:54:41 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 5 May 2010 08:54:41 +0200
Subject: [Freeswitch-users] run error after building in vs 2008!
In-Reply-To: <n2o47d63d921005042246m6e506ab8s403be7f167229611@mail.gmail.com>
References: <i2v1fdf92721004302253l7dbdf254p1ff2b3884ec5a4d4@mail.gmail.com>
	<l2o367751821005010612j46281f6fg17629c38bcd07f4@mail.gmail.com>
	<n2i1fdf92721005011131o221b6dd4sf483e00613963d5@mail.gmail.com>
	<x2z367751821005011229g3b6c217kbda766daac32a44d@mail.gmail.com>
	<E141A983-9EC5-4C62-8D17-675DC0A67009@jerris.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5576CE4D97@cooper>
	<k2y87f2f3b91005031727j1661d31vf76dc19d9e3354e5@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5576CE4DA1@cooper>
	<n2n1fdf92721005042219x95407e2cp9a0496b95dee3308@mail.gmail.com>
	<n2o47d63d921005042246m6e506ab8s403be7f167229611@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C55777E1C40@cooper>

This is the link about FS/FreeTDM: http://wiki.sangoma.com/wanpipe-api-freetdm

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r ovvenkat
Skickat: den 5 maj 2010 07:47
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] run error after building in vs 2008!

Hi to all,

We have developed a IVR Application in freeSwitch SDK. Now , we want to  deploye it on the production server. So, we  I bought Sangoma A101 T1/E1 card. Now we are facing  a new problem like

FreeSwitch wont support "Europe ISDN" connection. their technical support guys are saying that, we need to buy a additional software called  "NetBorderExpress" to overcome this problem. Is this true? why its so?

Regards
Venkatesan OV.
!DSPAM:4be1079832931102718702!
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From msc at freeswitch.org  Tue May  4 23:55:42 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 4 May 2010 23:55:42 -0700
Subject: [Freeswitch-users] problem in SDK while deploying app on the
	production server
In-Reply-To: <k2y47d63d921005042303oef5a58d9kd70fddf5ea42306d@mail.gmail.com>
References: <k2y47d63d921005042303oef5a58d9kd70fddf5ea42306d@mail.gmail.com>
Message-ID: <y2l87f2f3b91005042355s34b3740bt366f0c60d0dd73e7@mail.gmail.com>

On Tue, May 4, 2010 at 11:03 PM, ovvenkat <ovvenkatesan at gmail.com> wrote:

> Hi Brian,
>
> thanks you very much for your quick reply. Can you send me the wiki link of
> the same.
>
> Start here:
http://wiki.sangoma.com/wanpipe-freeswitch

-MC


>
> Regards,
> Venkat.
>
> On Wed, May 5, 2010 at 11:22 AM, Brian West <brian at freeswitch.org> wrote:
>
>> That's a bull, I was told today it does support euroisdn and the wiki
>> seems to agree with me.
>>
>> /b
>>
>> Sent from my iPad
>>
>> On May 5, 2010, at 12:46 AM, ovvenkat <ovvenkatesan at gmail.com> wrote:
>>
>> Hi to all,
>>
>> We have developed a IVR Application in freeSwitch SDK. Now , we want to
>> deploye it on the production server. So, we  I bought Sangoma A101 T1/E1
>> card. Now we are facing  a new problem like
>>
>> *FreeSwitch wont support "Europe ISDN" connection*. their technical
>> support guys are saying that, we need to buy a additional software called  "
>> *NetBorderExpress*" to overcome this problem. Is this true? why its so?
>>
>> Regards
>> Venkatesan OV.
>>
>> _______________________________________________
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> Regards
> Venkatesan OV.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ovvenkatesan at gmail.com  Wed May  5 00:24:11 2010
From: ovvenkatesan at gmail.com (ovvenkat)
Date: Wed, 5 May 2010 12:54:11 +0530
Subject: [Freeswitch-users] problem in SDK while deploying app on the
	production server
In-Reply-To: <y2l87f2f3b91005042355s34b3740bt366f0c60d0dd73e7@mail.gmail.com>
References: <k2y47d63d921005042303oef5a58d9kd70fddf5ea42306d@mail.gmail.com>
	<y2l87f2f3b91005042355s34b3740bt366f0c60d0dd73e7@mail.gmail.com>
Message-ID: <u2j47d63d921005050024y2d552c5amea07a06f7e46d78c@mail.gmail.com>

thanks mic. I got it. :)

On Wed, May 5, 2010 at 12:25 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Tue, May 4, 2010 at 11:03 PM, ovvenkat <ovvenkatesan at gmail.com> wrote:
>
>> Hi Brian,
>>
>> thanks you very much for your quick reply. Can you send me the wiki link
>> of the same.
>>
>> Start here:
> http://wiki.sangoma.com/wanpipe-freeswitch
>
> -MC
>
>
>>
>> Regards,
>> Venkat.
>>
>> On Wed, May 5, 2010 at 11:22 AM, Brian West <brian at freeswitch.org> wrote:
>>
>>> That's a bull, I was told today it does support euroisdn and the wiki
>>> seems to agree with me.
>>>
>>> /b
>>>
>>> Sent from my iPad
>>>
>>> On May 5, 2010, at 12:46 AM, ovvenkat <ovvenkatesan at gmail.com> wrote:
>>>
>>> Hi to all,
>>>
>>> We have developed a IVR Application in freeSwitch SDK. Now , we want to
>>> deploye it on the production server. So, we  I bought Sangoma A101 T1/E1
>>> card. Now we are facing  a new problem like
>>>
>>> *FreeSwitch wont support "Europe ISDN" connection*. their technical
>>> support guys are saying that, we need to buy a additional software called  "
>>> *NetBorderExpress*" to overcome this problem. Is this true? why its so?
>>>
>>> Regards
>>> Venkatesan OV.
>>>
>>> _______________________________________________
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> Regards
>> Venkatesan OV.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 

If you have come to help me, you are wasting your time.
If you have come to because your liberation is bound up in mine, we can work
together.


Regards
Venkatesan OV.
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From david.ponzone at gmail.com  Wed May  5 00:26:35 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Wed, 5 May 2010 09:26:35 +0200
Subject: [Freeswitch-users] Annex B
In-Reply-To: <r2j87f2f3b91005042351rfda2eb02w407283c35e6f8658@mail.gmail.com>
References: <01af01caebd6$c80c04c0$58240e40$@com>
	<F831BDC6-59B9-4539-B288-572504D5F605@freeswitch.org>
	<01c001caebda$7d9f7300$78de5900$@com>
	<911ACB36-F863-4C76-9467-B6A2FEDFC0E6@freeswitch.org>
	<68A2CF78-F6AB-4D70-8F49-E2EA362F8B99@avgs.ca>
	<139A3BAE-6616-44BF-8CA2-D8F685C1A767@freeswitch.org>
	<r2j87f2f3b91005042351rfda2eb02w407283c35e6f8658@mail.gmail.com>
Message-ID: <843FE445-D78D-47D4-9622-BBD5EDAF9F7B@gmail.com>

Guys,

I quickly documented this secret chanvar from Brian's stash as it was  
missing in the wiki:

http://wiki.freeswitch.org/wiki/Channel_Variables#sip_append_audio_sdp

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 05/05/2010 ? 08:51, Michael Collins a ?crit :

>
>
> On Tue, May 4, 2010 at 4:21 PM, Brian West <brian at freeswitch.org>  
> wrote:
> Not sure you want to search all I have in my logs :P
> Hehe, I'm dead sure that I don't. :D
>
> Ken, maybe you could document this on the wiki. We have a codecs  
> page as well as a g.729 page. Either one might be appropriate.
> -MC
>
>
> /b
>
> On May 4, 2010, at 6:16 PM, Mathieu Rene wrote:
>
>> Lets work on a spotlight module to search Brian's chat logs from  
>> freeswitch.org
>>
>> Mathieu Rene
>> Avant-Garde Solutions Inc
>> Office: + 1 (514) 664-1044 x100
>> Cell: +1 (514) 664-1044 x200
>> mrene at avgs.ca
>>
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From vfclists at googlemail.com  Wed May  5 03:51:28 2010
From: vfclists at googlemail.com (Frank Church)
Date: Wed, 5 May 2010 11:51:28 +0100
Subject: [Freeswitch-users] mod_xml_cdr error log is not working
In-Reply-To: <4C4850C4-3704-4646-B962-675FFF54B363@jerris.com>
References: <g2p7abab2411004291856s2c2d5018g103f33cf2be9dd4b@mail.gmail.com>
	<DD0949EC-0C93-4A8F-9EA4-C6A669AE93FF@jerris.com>
	<l2z7abab2411005030343xdbe1827coc7e531214aa78864@mail.gmail.com>
	<68DF5166-42A6-4316-9F96-3FD3CC6C9B62@jerris.com>
	<v2w7abab2411005031403v5e00a8ccx7f2f2a5dadef3b29@mail.gmail.com>
	<i2p7abab2411005031405o7438559bu9bc450aca3f93e35@mail.gmail.com>
	<4C4850C4-3704-4646-B962-675FFF54B363@jerris.com>
Message-ID: <t2h7abab2411005050351p7b40b030x33f119efcae9e233@mail.gmail.com>

I can confirm that if the logging directory is changed from the default, it
will not be used unless log-http-and-disk is set. It will still log to the
xml_cdr directory if that exists, even though it is not what you have
specified.

On 4 May 2010 23:12, Michael Jerris <mike at jerris.com> wrote:

> so is this actually a bug as well?
>
> Mike
>
> On May 3, 2010, at 5:05 PM, Frank Church wrote:
>
> PS. Brian has updated the config samples,
> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-May/057475.html
>
> On 3 May 2010 22:03, Frank Church <vfclists at googlemail.com> wrote:
>
>>
>>
>> I think it is a bug because the wiki states that log-http-and-disk Default
>> behaviour is to write either HTTP or Disk on HTTP failure. Setting this to
>> true will write to both HTTP and Disk regardless (handy for realtime +
>> reconciliation later if required) true
>> But it wasn't writing the HTTP connection errors to the disk until I added
>> log-http-and-disk to the configuration. log-http-and-disk is for writing the
>> CDR itself, but logging HTTP errors did not work until it was added
>>
>>
>> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-May/057475.html
>>
>> On 3 May 2010 21:14, Michael Jerris <mike at jerris.com> wrote:
>>
>>> would you mind making sure this gets documented on the wiki, and, if it
>>> is not in the sample configuration, send us a patch for that via
>>> http://jira.freeswitch.org ?
>>>
>>> Mike
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Frank Church

=======================
http://devblog.brahmancreations.com
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From mark at mdsh.com  Wed May  5 05:28:31 2010
From: mark at mdsh.com (Mark Himsley)
Date: Wed, 05 May 2010 13:28:31 +0100
Subject: [Freeswitch-users] alpha-numeric password
In-Reply-To: <AANLkTimWn0SSDF8gbwtu4CL6MSYaQ8fQ61p4ixUONm9v@mail.gmail.com>
References: <4BE00C5A.6090201@mdsh.com>
	<AANLkTimWn0SSDF8gbwtu4CL6MSYaQ8fQ61p4ixUONm9v@mail.gmail.com>
Message-ID: <4BE1646F.2060601@mdsh.com>

Thanks for your unequivocal answer.

FS was sent a HUP signal with `sudo /etc/init.d/freeswitch reload` with
the standard FS apt-get install Ubuntu server 9.10.

I now realise that this command was not reloading
${FREESWITCH_CONF}/directory/default/201.xml and I needed to do a
restart instead.

I can see that I need to get to the freeswitch command line (like *'s
-r) for better configuring/debugging so I'll look into running FS in
screen, hopefully without destroying the Ubuntu standard init script too
much.

Thanks. Your unequivocal answer made me do a lot more debugging :-)


On 04/05/2010 17:48, Anthony Minessale wrote:
> There is no such limitation
> did you do reloadxml and or restart FS after you changed the configuration?
> 
> 
> On Tue, May 4, 2010 at 7:00 AM, Mark Himsley <mark at mdsh.com
> <mailto:mark at mdsh.com>> wrote:
> 
>     Hi,
> 
>     I'm very new to Freeswitch - I want to move from Asterisk to Freeswitch
>     for my home VoIP switch.
> 
>     In Asterisk sip.conf I have defined extensions like this
> 
>     [201]
>     type=friend
>     username=201
>     secret=mypass
>     <...SNIP...>
> 
>     If I create ${FREESWITCH_CONF}/directory/default/201.xml starting
>     like this:
> 
>     <include>
>      <user id="201">
>        <params>
>          <param name="password" value="mypass"/>
>     <...SNIP...>
> 
>     and just change the server the phone connects to to be my new freeswitch
>     server then the phone cannot authentcate:
> 
>     2010-05-04 11:48:28.936358 [WARNING] sofia_reg.c:1030 SIP auth failure
>     (REGISTER) on sofia profile 'internal' for [201 at 10.0.0.10
>     <mailto:201 at 10.0.0.10>] from ip
>     10.0.0.228
> 
>     But if I set the value element of the password param to a number and
>     also change the password in the phone to that number then the phone will
>     authenticate.
> 
>     Does freeswitch only allow numeric passwords for SIP phones? I can't see
>     a definition on the freeswitch web site for what is valid for a
>     password, and all my Googling has failed too :-(
> 
>     I'd _like_ to keep the passwords unchanged, but I can live with changing
>     them if needed. I was just wondering if I missed something on the
>     freeswitch web site.
> 
>     Thanks in advance.
> 
>     --
>     Mark
> 
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> <mailto:sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From woodydickson at gmail.com  Wed May  5 06:27:26 2010
From: woodydickson at gmail.com (Woody Dickson)
Date: Wed, 5 May 2010 21:27:26 +0800
Subject: [Freeswitch-users] freeSWITCH segfault on openwrt
Message-ID: <i2ha44688a1005050627n8739d2bcl280c67dddca79866@mail.gmail.com>

Hi,

I manage to get freeSWITCH to compile and run on openwrt, but then
freeSWITCH keeps crashing when the number of calls reaches 25
consistently.

Content of the core file returns nothing when opened with gdb.

Does anyone have any clue as to why this is happening and how to fix it?

thanks,
woody


From david.ponzone at gmail.com  Wed May  5 07:12:48 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Wed, 5 May 2010 16:12:48 +0200
Subject: [Freeswitch-users] Different codecs for different devices?
In-Reply-To: <i2xebbc2d551005041450qfec08896l527fa2ee3c8a1639@mail.gmail.com>
References: <009601caeb8d$cd5953e0$680bfba0$@com>
	<010f01caeba7$6ea4a470$4beded50$@com>
	<AANLkTiluKFA2nqpDr4lUKujb2YFw5vAawYQRjlf7JOc6@mail.gmail.com>
	<i2xebbc2d551005041450qfec08896l527fa2ee3c8a1639@mail.gmail.com>
Message-ID: <8F430902-764D-4D51-9F3A-E55F9D93F073@gmail.com>

Guys,

I added this variable to the wiki:

http://wiki.freeswitch.org/wiki/Channel_Variables#variable_xxxx

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 04/05/2010 ? 23:50, Jo?o Mesquita a ?crit :

> I think you can also use another variable on the A leg SDP to make  
> it easier on the regex. ep_codec_string.
>
> This will contain the parsed version of the A leg SDP and that's  
> what I use most of the times for these tricks. Also, can you  
> document that? :-)
>
> Regards,
> Jo?o Mesquita
>
>
> On Tue, May 4, 2010 at 1:52 PM, Anthony Minessale <anthony.minessale at gmail.com 
> > wrote:
> you can regex the SDP for something specific and use that to set the  
> absolute_codec_string in a standalone extension that comes first in  
> the stack and has continue=true on it
>
>
>
> On Tue, May 4, 2010 at 11:32 AM, Ken Fulmer <kenfulmer at icstechnologysolutions.com 
> > wrote:
> Just in case others are interested, this is possible with the  
> following parameters:
>
>
> SIP Profile:
>
>
>     <param name="inbound-late-negotiation" value="true"/>
>
>
> Dial-Plan:
>
>
>       <action application="set" data="inherit_codec=true"/>
>
>
> The codec negotiated in Call Leg B, is forced onto Call Leg A. This  
> is possible due to the ?inbound late negotiation? parameter in the  
> sip profile.
>
>
> Hope this helps someone else.
>
>
> Ken
>
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Ken Fulmer
> Sent: Tuesday, May 04, 2010 8:29 AM
>
>
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Different codecs for different devices?
>
>
> This may be a far out question but here goes:
>
>
> Our fax machines that are connected to ATA?s need to be g.711 on  
> both Call Leg A and Call Leg B (in either direction).
>
>
> However, our phones need to be g.711 on the internal side that faces  
> our PBX. On the side facing our SIP provider, the calls should be  
> transcoded to g.729. Inbound the calls to phones (not ATA?s) need to  
> be g.729 on the inbound call leg and g.711 on the outbound call leg.
>
>
> I realize we can set inbound / outbound codec preferences in single  
> or multiple profiles and we can set the ?absolute_codec_string?  
> value for the outbound call leg.
>
>
> However, I?ve yet to determine how to set the inbound call leg?s  
> codec differently depending on the device. Is this possible?
>
>
> Thanks,
>
>
> Ken Fulmer
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From azmanminha at yahoo.com  Tue May  4 23:43:14 2010
From: azmanminha at yahoo.com (Azman Minha)
Date: Tue, 4 May 2010 23:43:14 -0700 (PDT)
Subject: [Freeswitch-users] server to server jingle mod_dingaling
Message-ID: <498383.40262.qm@web38908.mail.mud.yahoo.com>

Hi,

 I use jingle protocol (my XMPP is jabberd) to bridge the call to FS.
 So far I managed to see the jingle XML packet sent from my client at FS.

 However,that is it.Nothing happen after that at FS.FS just do nothing.

 What could go wrong?What should I do next?

 What is missing in my server jingle profiles.

 <!-- Component (Server to Server Login) --
  <!-- to use this profile take the x- away from the open and close tags so its <profile and </profile --
  <profile type="component"
    <param name="name" value="sip.myjabber.com.my"/
    <param name="password" value="secret"/
    <param name="dialplan" value="XML"/
    <param name="context" value="public"/
    <!--param name="rtp-ip" value="$${bind_server_ip}"/--
    <param name="rtp-ip" value="auto"/
    <param name="server" value="xx.xx.xx.xx"/
    <!-- disable to trade async for more calls --
    <param name="use-rtp-timer" value="true"/
    <!-- "_auto_" means the extension will be automaticly set to the called jid --
    <param name="exten" value="_auto_"/
    <param name="vad" value="both"/
    <!--param name="avatar" value="/path/to/tiny.jpg"/--
    <!--If you have ODBC support and a working dsn you can use it instead of SQLite--
    <!--param name="odbc-dsn" value="dsn:user:pass"/--
    <param name="candidate-acl" value="wan.auto"/
    <param name="auto-login" value="true"/
  </profile


 <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"
  <settings
    <param name="debug" value="1"/
    <param name="codec-prefs" value="PCMU"/
  </settings

  <X-PRE-PROCESS cmd="include" data="../jingle_profiles/*.xml"/

 </configuration

 Thanks.


      


From steve.d.ward at gmail.com  Wed May  5 07:33:07 2010
From: steve.d.ward at gmail.com (Steven Ward)
Date: Wed, 5 May 2010 10:33:07 -0400
Subject: [Freeswitch-users] Method to force a transfer of A-Leg
In-Reply-To: <m2p4ea6e8f21005041253y8dcd7190pb3e2c00c49ce4ecf@mail.gmail.com>
References: <p2w4ea6e8f21004210709w54a18d7ao6bb9adb2acaaca8a@mail.gmail.com>
	<EC5C7D5E-DC08-4A5E-B93B-F9D42EF9EE5E@jerris.com>
	<w2m4ea6e8f21005041155x6a4ccbbgc64cdbb59114c790@mail.gmail.com>
	<AANLkTilzLGTOxy7aKWW669j3h-_ayis5cu6PPWOli3fY@mail.gmail.com>
	<m2p4ea6e8f21005041253y8dcd7190pb3e2c00c49ce4ecf@mail.gmail.com>
Message-ID: <n2h4ea6e8f21005050733k3d36ccd0zd56018d40c806047@mail.gmail.com>

So the way I've tried interrupting a transfer of the type described in this
thread is by using uuid_deflect, but I have a strange problem with that.

Most of the time it works just fine.  But sometimes the deflected call
dies.

I've done sip traces and looked at the log, but it's a bit baffling to me
why it doesn't work all the time.

>From a lua script, I do a uuid_deflect on the a-leg to
sip:[MyNewDestinationURI], where MyNewDestinationURI gets routed to a
different destination on my FS box.

So when I do the uuid_deflect, I see the a-leg's device (let's call that
Phone A) get the REFER, the a-leg hangs up, and a new call comes back into
FS from device Phone A to the new destination.

Like I said, most of the time this works, so FS sends Session Progress and
eventually an OK for that INVITE.

But sometimes FS simply sends a BYE and a 486 Busy Here in response to the
INVITE.  (The cause for the BYE is "BLIND_TRANSFER").

Doing a diff on the logs for when the uuid_deflect works and when it doesn't
work, the only difference I see is as follows:
      * when the deflected call does get answered successfully, I see the
messages:

[DEBUG] sofia.c:5831 IP 10.244.16.26 Approved by acl "domains[]". Access
Granted.
[NOTICE] switch_channel.c:675 New Channel
sofia/internal/25327 at 10.244.16.23[82e4521a-5848-11df-b467-cd6b5075e2a0]

   (above, 10.244.16.26 is the IP of the SIP router for device Phone A)

      * when the deflected call does NOT get answered, but is sent a BYE and
Busy Here, I notice the above messages do NOT appear in the log at all.

I'm on what was latest git pull as of yesterday morning.  Is this something
where I should pastebin the logs and sip traces?

And the Busy Here seems to strange to me, because the endpoint on FS it's
getting sent to is dialplan that does a 'fifo in' - and no one else is
calling to execute that dialplan when I ran these tests, so it's not really
busy as far as I understand.

Thanks again for the help.  Any feedback is much appreciated.





On Tue, May 4, 2010 at 12:53 PM, Steven Ward <steve.d.ward at gmail.com> wrote:

> At the moment, I'm not using an ivr.  The transfer is done by the phone (so
> when I hit the Transfer button, the phone puts a-leg on hold and I dial a
> new call; I hit the Transfer button again and the Polycom does the REFER w/
> REPLACES and hangs up on a-leg and the new destination).
>
> (Makes me wish the phone didn't come with its own built-in Transfer feature
> with accompanying button and I could have FS handle everything... :-)
>
>
> On Tue, May 4, 2010 at 3:40 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> Could you just collect the digits from the user in the ivr, transfer him
>> back to the conference.
>> then originate the call separately with the collected digits to enter the
>> conference once it answers?
>>
>>
>> On Tue, May 4, 2010 at 1:55 PM, Steven Ward <steve.d.ward at gmail.com>wrote:
>>
>>> Sorry I haven't provided feedback any sooner...
>>>
>>> I've been looking at uuid_transfer and uuid_deflect, and I'm still having
>>> a tough time getting the desired effect for my situation.
>>>
>>> I hope these details make things a bit more clear.
>>>
>>> The thing is, this script is monitoring transfers that are done by a
>>> Polycom; the transfer starts as attended, but then is converted to a blind
>>> xfer (so the xfer is ringing, and I hit the Transfer button again to exit
>>> out of the transfer).  So the Polycom sends a REFER to the a-leg w/ a
>>> REPLACES param.
>>>
>>> My script then turns on, sleeps for a bit, and then checks out the
>>> a-leg.  If the a-leg hasn't been answered, I'd like the script to cancel
>>> that transfer and move the leg somewhere else on the box.
>>>
>>> The thing is, the a-leg isn't bridged yet because it hasn't been answered
>>> (destination does not even send early media).  And the a-leg doesn't even
>>> know in its channel variables what channel it's trying to ring, because of
>>> the way the transfer works.  The a-leg is just waiting for the transfer to
>>> complete.  As Tony graciously described to me on irc one time, until the
>>> destination answers (or sends early media), the channel that dialed out to
>>> do the transfer remains up in a zombified state.  And once the destination
>>> answers, then the zombified channel drops out and the a-leg is bridged to
>>> the destination.
>>>
>>> Incidentally, I do see that FS keeps track of the "zombie" channel in the
>>> a-leg's *att_xfer_kill_uuid* channel variable.
>>>
>>> The impact of this to what I'm trying to do is that uuid_transfer doesn't
>>> work for taking over the call.  I can uuid_transfer the a-leg, but the
>>> original destination of the transfer keeps ringing and the
>>> att_xfer_kill_uuid channel stays up.  So now it's as if the a-leg is ringing
>>> two destinations, and the original destination never stops ringing.
>>>
>>> Is there any way I can find out what channel the a-leg is trying to get
>>> connected to via att_xfer_kill_uuid?  Or is there any way I can truly cancel
>>> that attended xfer and take control of the a-leg?
>>>
>>> I see that uuid_deflect can be used to send a REFER and send it off-box,
>>> but I'd like to simply send the a-leg through dialplan on-box.  (And when
>>> I've tried using uuid_deflect to REFER it back to another destination on my
>>> same box, I *sometimes* get a busy for a reason I haven't been able to debug
>>> yet.)
>>>
>>> So, any thoughts on controlling the Polycom
>>> attended-transfer-turned-blind :) would be much appreciated.
>>>
>>> This is for an application that replicates the features of an operator
>>> console that is currently in use - that's why I need to control calls so
>>> maniacally.  Once I get this polished off, I look forward to putting the
>>> final results onto the wiki as an example application.
>>>
>>> Thanks again for all the help I've already received on irc and the list
>>> in the past, and thanks for any comments on this scenario...
>>>
>>> - Steve
>>> (sward_)
>>> steve.d.ward at gmail.com
>>>
>>>
>>> On Sat, Apr 24, 2010 at 2:47 PM, Michael Jerris <mike at jerris.com> wrote:
>>>
>>>> just uuid_transfer the a leg.
>>>>
>>>> On Apr 21, 2010, at 10:09 AM, Steven Ward wrote:
>>>>
>>>> Hello all,
>>>>
>>>> I have a lua script running that checks the state of a call between A
>>>> and B - the call between A and B was set up through a Polycom's attended
>>>> transfer feature, so A itself didn't necessarily execute the bridge to B.
>>>>
>>>> A gets early media from B.  After a certain amount of time, if A is
>>>> still getting early media, I want the script to end that call between A and
>>>> B, and send A through some specific dialplan.
>>>>
>>>> How does uuid_transfer work for this goal?  I'd like to kill B and move
>>>> A to specific dialplan.  I don't want B to be in the transfer at all.
>>>>
>>>> I know there may be many possibilities here; I'm just wondering if
>>>> anyone can recommend something.  Thanks.
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <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-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From freeswitch at bsdjournal.net  Wed May  5 08:01:41 2010
From: freeswitch at bsdjournal.net (Bryan Vyhmeister)
Date: Wed, 5 May 2010 11:01:41 -0400
Subject: [Freeswitch-users] Sangoma prid and multiple PRIs with FreeSWITCH
Message-ID: <x2y7e9692c31005050801ud0a264f6u709549a8790e10b0@mail.gmail.com>

I setup a FreeSWITCH pbx with a Sangoma A102D card, the latest FreeSWITCH as
of yesterday, and Wanpipe 3.5.11. I have found that I cannot use both PRI
ports with Sangoma's smg_prid. The first one always works great but the
second one never works. I get the following error:

2010-05-05 09:42:02.573743 [CRIT] ozmod_sangoma_boost.c:257 SPAN is not
online.
2010-05-05 09:42:02.573743 [ERR] switch_ivr_originate.c:2480 Cannot create
outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]

When I start FreeSWITCH up I get this error on the console right after all
the channels are activated.

2010-05-05 07:25:59.414444 [ERR] ozmod_sangoma_boost.c:1534 Error: Opening
MCON Socket [-1] Address already in use

The configs are below:

openzap.conf
[span wanpipe 1]
name => 1
trunk_type =>t1
b-channel => 1:1-23
[span wanpipe 2]
name => 2
trunk_type =>t1
b-channel => 2:1-23

openzap.conf.xml
<configuration name="openzap.conf" description="OpenZAP Configuration">
  <settings>
    <param name="debug" value="0"/>
  </settings>
  <boost_spans>
<span id="1">
        <param name="node" value="cpe"/>
        <param name="switch" value="ni2"/>
<param name="dialplan" value="XML"/>
 <param name="context" value="public"/>
</span>
 <span id="2">
        <param name="node" value="cpe"/>
        <param name="switch" value="ni2"/>
 <param name="dialplan" value="XML"/>
<param name="context" value="public"/>
  </boost_spans>
  <analog_spans>
  </analog_spans>
</configuration>

I'm dialing with this dial string.

        <action application="bridge"
data="{origination_caller_id_number=${outbound_caller_id_number}}openzap/2/a/$1"/>

I am certain the rest of this config works fine because the other PRI works
great. I have also attempted to use FreeSWITCH without Sangoma's smg_prid by
specifying a d-channel in openzap.conf. That seems to work but calls tend to
drop frequently. I would rather just use Sangoma's smg_prid. I also
attempted a hybrid config where FreeSWITCH worked without smg_prid on the
first PRI and the second PRI was handled by smg_prid. That config worked
except for the dropped calls that happen with using FreeSWITCH to manage the
d-channel.

My question is how can I get two PRI's working with Sangoma's smg_prid?
Thank you.

Bryan
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From jmesquita at freeswitch.org  Wed May  5 08:09:05 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 5 May 2010 12:09:05 -0300
Subject: [Freeswitch-users] Different codecs for different devices?
In-Reply-To: <8F430902-764D-4D51-9F3A-E55F9D93F073@gmail.com>
References: <009601caeb8d$cd5953e0$680bfba0$@com>
	<010f01caeba7$6ea4a470$4beded50$@com>
	<AANLkTiluKFA2nqpDr4lUKujb2YFw5vAawYQRjlf7JOc6@mail.gmail.com>
	<i2xebbc2d551005041450qfec08896l527fa2ee3c8a1639@mail.gmail.com>
	<8F430902-764D-4D51-9F3A-E55F9D93F073@gmail.com>
Message-ID: <p2lebbc2d551005050809oc5491108h4e877d635334d685@mail.gmail.com>

You are the man. Thank you for documenting those. I have a few on my list
that I never seem to find the freaking time to actually do them.

Regards,

JM


On Wed, May 5, 2010 at 11:12 AM, David Ponzone <david.ponzone at gmail.com>wrote:

> Guys,
>
> I added this variable to the wiki:
>
> http://wiki.freeswitch.org/wiki/Channel_Variables#variable_xxxx
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 04/05/2010 ? 23:50, Jo?o Mesquita a ?crit :
>
> I think you can also use another variable on the A leg SDP to make it
> easier on the regex. ep_codec_string.
>
> This will contain the parsed version of the A leg SDP and that's what I use
> most of the times for these tricks. Also, can you document that? :-)
>
> Regards,
> Jo?o Mesquita
>
>
> On Tue, May 4, 2010 at 1:52 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> you can regex the SDP for something specific and use that to set the
>> absolute_codec_string in a standalone extension that comes first in the
>> stack and has continue=true on it
>>
>>
>>
>> On Tue, May 4, 2010 at 11:32 AM, Ken Fulmer <
>> kenfulmer at icstechnologysolutions.com> wrote:
>>
>>>  Just in case others are interested, this is possible with the following
>>> parameters:
>>>
>>>
>>> SIP Profile:
>>>
>>>
>>>     <param name="inbound-late-negotiation" value="true"/>
>>>
>>>
>>> Dial-Plan:
>>>
>>>
>>>       <action application="set" data="inherit_codec=true"/>
>>>
>>>
>>> The codec negotiated in Call Leg B, is forced onto Call Leg A. This is
>>> possible due to the ?inbound late negotiation? parameter in the sip profile.
>>>
>>>
>>> Hope this helps someone else.
>>>
>>>
>>> Ken
>>>
>>>
>>>
>>>
>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ken Fulmer
>>> *Sent:* Tuesday, May 04, 2010 8:29 AM
>>>
>>> *To:* freeswitch-users at lists.freeswitch.org
>>> *Subject:* [Freeswitch-users] Different codecs for different devices?
>>>
>>>
>>>
>>> This may be a far out question but here goes:
>>>
>>>
>>> Our fax machines that are connected to ATA?s need to be g.711 on both
>>> Call Leg A and Call Leg B (in either direction).
>>>
>>>
>>> However, our phones need to be g.711 on the internal side that faces our
>>> PBX. On the side facing our SIP provider, the calls should be transcoded to
>>> g.729. Inbound the calls to phones (not ATA?s) need to be g.729 on the
>>> inbound call leg and g.711 on the outbound call leg.
>>>
>>>
>>> I realize we can set inbound / outbound codec preferences in single or
>>> multiple profiles and we can set the ?absolute_codec_string? value for the
>>> outbound call leg.
>>>
>>>
>>> However, I?ve yet to determine how to set the inbound call leg?s codec
>>> differently depending on the device. Is this possible?
>>>
>>>
>>> Thanks,
>>>
>>>
>>> Ken Fulmer
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <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-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jerry.richards at teotech.com  Wed May  5 08:17:23 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 5 May 2010 08:17:23 -0700
Subject: [Freeswitch-users] ** Feature Code
Message-ID: <BEDC45F3CA404E8FAC07AEF3EA7979B5@greyhawk.tonecommander.com>


As you know, dialing **<number> picks up a ringing call (i.e. Directed Call
Pickup).  However, I noticed yesterday that it will also take a call away
from another phone even after it is already connected.  Here is the
scenario:

1) 5382 calls 5403.
2) 5403 answers and two-way talk path.
3) 5381 dials **5403
4) 5403 loses call.
5) 5382 and 5382 two-way talk path.

Is there a way to keep the Directed Call Pickup feature, but prevent taking
over the call after connect?

Thanks And Best Regards,
Jerry



From hungngm1987 at gmail.com  Wed May  5 08:23:38 2010
From: hungngm1987 at gmail.com (hung nguyen)
Date: Wed, 5 May 2010 22:23:38 +0700
Subject: [Freeswitch-users] How to make scheduled calls ?
Message-ID: <o2rd9f1e4661005050823q704bf7feje4a19d216eef98fe@mail.gmail.com>

Hi list.

I want config FS to make automatic calls follow a fixed schedule.
Example: In fixed time in the future, FS will call a fixed user and play a
fixed music file for this user.

I don't want to use crontab to do this because i want to deploy large
schedules.

Thank to suggestions.
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From jerry.richards at teotech.com  Wed May  5 08:23:56 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 5 May 2010 08:23:56 -0700
Subject: [Freeswitch-users] ** Feature Code
Message-ID: <F7CB14D09E8B47D58D44C1B63A94985E@greyhawk.tonecommander.com>

Sorry there was a typo in my scenario. Here is the correction:

1) 5382 calls 5403.
2) 5403 answers and two-way talk path.
3) 5381 dials **5403
4) 5403 loses call.
5) 5381 and 5382 two-way talk path.

Thanks,
Jerry

-----Original Message-----
From: Jerry Richards [mailto:jerry.richards at teotech.com] 
Sent: Wednesday, May 05, 2010 8:17 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: ** Feature Code


As you know, dialing **<number> picks up a ringing call (i.e. Directed Call
Pickup).  However, I noticed yesterday that it will also take a call away
from another phone even after it is already connected.  Here is the
scenario:

1) 5382 calls 5403.
2) 5403 answers and two-way talk path.
3) 5381 dials **5403
4) 5403 loses call.
5) 5382 and 5382 two-way talk path.

Is there a way to keep the Directed Call Pickup feature, but prevent taking
over the call after connect?

Thanks And Best Regards,
Jerry



From msc at freeswitch.org  Wed May  5 08:31:07 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 May 2010 08:31:07 -0700
Subject: [Freeswitch-users] FreeSWITCH Community Conference Call May 5
Message-ID: <t2x87f2f3b91005050831n79a4fddcvb2dd33c49522d409@mail.gmail.com>

Hello all!

Our weekly conference call will be starting in about 90 minutes. (1pm
Eastern/10am Pacific) The agenda is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2010_05_05

I have only a few things to discuss so please bring your questions and
thoughts. Multilingual community members are especially needed today so that
we can talk about translating phrase files. If you have any other things to
discuss please by all means bring them. Rupa is traveling this week and
won't be available this week for one of his killer presentations. :)

Thanks everyone and hope to talk to you in a few...

-Michael
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From mike at jerris.com  Wed May  5 08:42:30 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 5 May 2010 11:42:30 -0400
Subject: [Freeswitch-users] Suggestions for creating diff's in git...
In-Reply-To: <v2q7abab2411005031459o36c05f43x4f8ce823a08d9b54@mail.gmail.com>
References: <549CFEF87AEDE841A38E9D15EAB4C04C5576CE4D98@cooper>
	<8E40D6E4-05BE-49D4-8BED-BECA792D6B7A@freeswitch.org>
	<549CFEF87AEDE841A38E9D15EAB4C04C5576CE4D99@cooper>
	<v2q7abab2411005031459o36c05f43x4f8ce823a08d9b54@mail.gmail.com>
Message-ID: <E6DB5478-62E2-4BDF-92A2-75F3CBA84571@jerris.com>

http://www.kernel.org/pub/software/scm/git/docs/git-format-patch.html

git format patch origin

Mike

On May 3, 2010, at 5:59 PM, Frank Church wrote:

> There are options in git for handling CR/LF between Windows and Linux files. Search Google for 'git crlf'
> 
> On 3 May 2010 22:06, Peter Olsson <peter.olsson at visionutveckling.se> wrote:
> Thanks,
> 
> Though, my problem is that I mostly work on Windows, and when piping the output to a file I think it messes up LF/CRLF. I thought if there was another way of doing it - since I can't find a way to output it directly to a file, without messing up CRLF.
> 
> I guess I have to start using Linux more :)
> 
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Brian West [brian at freeswitch.org]
> Skickat: den 3 maj 2010 22:50
> Till: freeswitch-users at lists.freeswitch.org
> ?mne: Re: [Freeswitch-users] Suggestions for creating diff's in git...
> 
> git diff usually works well.
> 
> /b
> 
> On May 3, 2010, at 3:43 PM, Peter Olsson wrote:
> 
> > I've been using SVN for a couple of years, but I'm quite new to git (as new as the FS project).
> >
> > I haven't yet found a good (easy) way to create diff's, to send up to jira cases, does anyone have a good tutorial for this?
> >
> > I've made some patches, but I haven't submitted them yet because of this.
> >
> > Regards,
> >
> > Peter
> 

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From mike at jerris.com  Wed May  5 08:52:05 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 5 May 2010 11:52:05 -0400
Subject: [Freeswitch-users] How to monitor find out which events are
	currently monitored, set only a particular group of events?
In-Reply-To: <j2zfe67a3341005032105z4452c3c1n36a1ccc7e0793213@mail.gmail.com>
References: <k2w7abab2411005031105n726c11dx434981234cce49f8@mail.gmail.com>
	<j2zfe67a3341005032105z4452c3c1n36a1ccc7e0793213@mail.gmail.com>
Message-ID: <0374D015-9A33-4FC0-901B-D18862543B83@jerris.com>

This shouldn't be that hard to add.  why not try making a patch and providing it as a contribution?

Mike

On May 4, 2010, at 12:05 AM, Nagalenoj H. wrote:

> I too didn't find any commands to get the registered events. It will be helpful, if there is some way.
> 
> On Mon, May 3, 2010 at 11:35 PM, Frank Church <vfclists at googlemail.com> wrote:
> 
> Is there a way of getting of list the event types that are being monitored by your connection to Freeswitch, other than those keeping a record of those you've added from your own end?
> 
> I want to add some without deleting what is present with the first event plain XXXX or  filter Event-Name XXXX command

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From mike at jerris.com  Wed May  5 08:54:28 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 5 May 2010 11:54:28 -0400
Subject: [Freeswitch-users] DTMF events
In-Reply-To: <CEAC61B68CDA414B80B82DA24CDF88AA@MOBILEE1705>
References: <CEAC61B68CDA414B80B82DA24CDF88AA@MOBILEE1705>
Message-ID: <C4562F4B-BB10-4EED-B546-F973BAC69F0B@jerris.com>

I strongly suggest as I said on irc that a softphone that only does inband dtmf is the absolutely wrong thing to do.  You would be better off not supporting dtmf at all than doing inband instead of 2833.

Mike

On May 4, 2010, at 2:48 AM, Madovsky wrote:

> Hi,
>  
> is it possible to use min dtmf at 300 ?
> it's only to avoid to record again some dtmf files I did
> for a  phone keypad I currently trying to develop

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From ChristianDamianidis at globalive.com  Wed May  5 08:55:48 2010
From: ChristianDamianidis at globalive.com (Christian Damianidis)
Date: Wed, 5 May 2010 11:55:48 -0400
Subject: [Freeswitch-users] hanguphook arguments
Message-ID: <66EA3166EB339A4489B06286C0876A8A0EB894FA@mailserv.Globalive.local>

Looking at http://wiki.freeswitch.org/wiki/Example_Hangup_hook

I see there are some default arguments, however I want to pass in my
own.

 

This may be some weak javascript knowledge on my part, but when I do 

 

session.setHangupHook(on_hangup(myparam))

 

It executes the function (obviously as the brackets make it a function
call).

 

Any tips on how to do this, is it possible?

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From infos at madovsky.org  Wed May  5 08:59:37 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 5 May 2010 11:59:37 -0400
Subject: [Freeswitch-users] DTMF events
References: <CEAC61B68CDA414B80B82DA24CDF88AA@MOBILEE1705>
	<C4562F4B-BB10-4EED-B546-F973BAC69F0B@jerris.com>
Message-ID: <EF5891F90EF746E0880E09A33CCBF83B@MOBILEE1705>

Mike,

this email has been set before I went on IRC.

Regards

Franck
  ----- Original Message ----- 
  From: Michael Jerris 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, May 05, 2010 11:54 AM
  Subject: Re: [Freeswitch-users] DTMF events


  I strongly suggest as I said on irc that a softphone that only does inband dtmf is the absolutely wrong thing to do.  You would be better off not supporting dtmf at all than doing inband instead of 2833.


  Mike


  On May 4, 2010, at 2:48 AM, Madovsky wrote:


    Hi,

    is it possible to use min dtmf at 300 ?
    it's only to avoid to record again some dtmf files I did
    for a  phone keypad I currently trying to develop




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


  _______________________________________________
  FreeSWITCH-users mailing list
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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From dyatsin at sangoma.com  Wed May  5 09:11:26 2010
From: dyatsin at sangoma.com (David Yat Sin)
Date: Wed, 05 May 2010 12:11:26 -0400
Subject: [Freeswitch-users] run error after building in vs 2008!
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C55777E1C40@cooper>
References: <i2v1fdf92721004302253l7dbdf254p1ff2b3884ec5a4d4@mail.gmail.com>	<l2o367751821005010612j46281f6fg17629c38bcd07f4@mail.gmail.com>	<n2i1fdf92721005011131o221b6dd4sf483e00613963d5@mail.gmail.com>	<x2z367751821005011229g3b6c217kbda766daac32a44d@mail.gmail.com>	<E141A983-9EC5-4C62-8D17-675DC0A67009@jerris.com>	<549CFEF87AEDE841A38E9D15EAB4C04C5576CE4D97@cooper>	<k2y87f2f3b91005031727j1661d31vf76dc19d9e3354e5@mail.gmail.com>	<549CFEF87AEDE841A38E9D15EAB4C04C5576CE4DA1@cooper>	<n2n1fdf92721005042219x95407e2cp9a0496b95dee3308@mail.gmail.com>	<n2o47d63d921005042246m6e506ab8s403be7f167229611@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C55777E1C40@cooper>
Message-ID: <4BE198AE.7060701@sangoma.com>

Hi Peter,
No you do not have to purchase NetborderExpress to support Europe ISDN.

You can either use
1. Freeswitch + Freetdm + Libpri
or
2. Freeswitch + Freetdm + sangoma_pri stack.

Both options will support EuroISDN variant.

The sangoma_pri stack on Windows is still in testing phase and not 
production ready. We will have instructions on how to deploy it on the 
Sangoma Wiki in a couple hours. I will post the link to the Sangoma wiki 
page as soon as its ready.

*David Yat Sin, **BEng**, **Software Engineer**
*Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON 
L3R 9T3 Canada
t. 1 905 474 1990 x 119 | e. _dyatsin at sangoma.com 
<mailto:dyatsin at sangoma.com>_


On 5/5/2010 2:54 AM, Peter Olsson wrote:
>
> This is the link about FS/FreeTDM: 
> http://wiki.sangoma.com/wanpipe-api-freetdm
>
> /Peter
>
> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *F?r *ovvenkat
> *Skickat:* den 5 maj 2010 07:47
> *Till:* freeswitch-users at lists.freeswitch.org
> *?mne:* Re: [Freeswitch-users] run error after building in vs 2008!
>
> Hi to all,
>
> We have developed a IVR Application in freeSwitch SDK. Now , we want 
> to  deploye it on the production server. So, we  I bought Sangoma A101 
> T1/E1 card. Now we are facing  a new problem like
>
> *FreeSwitch wont support "Europe ISDN" connection*. their technical 
> support guys are saying that, we need to buy a additional software 
> called  "*NetBorderExpress*" to overcome this problem. Is this true? 
> why its so?
>
> Regards
> Venkatesan OV.
>
> !DSPAM:4be1079832931102718702!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    
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From anthony.minessale at gmail.com  Wed May  5 09:32:51 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 5 May 2010 11:32:51 -0500
Subject: [Freeswitch-users] ** Feature Code
In-Reply-To: <F7CB14D09E8B47D58D44C1B63A94985E@greyhawk.tonecommander.com>
References: <F7CB14D09E8B47D58D44C1B63A94985E@greyhawk.tonecommander.com>
Message-ID: <j2l191c3a031005050932w59fbc00chc20e8bf08dda70dd@mail.gmail.com>

not without a patch to mod_intercept.
I made one in revision 58fe45a


intercept_unbridged_only=true should make it skip intercepting calls which
are bridged.
keep in mind that ** thing is just an example it's intercept app and how you
call it that makes it happen.



On Wed, May 5, 2010 at 10:23 AM, Jerry Richards
<jerry.richards at teotech.com>wrote:

> Sorry there was a typo in my scenario. Here is the correction:
>
> 1) 5382 calls 5403.
> 2) 5403 answers and two-way talk path.
> 3) 5381 dials **5403
> 4) 5403 loses call.
> 5) 5381 and 5382 two-way talk path.
>
> Thanks,
> Jerry
>
> -----Original Message-----
> From: Jerry Richards [mailto:jerry.richards at teotech.com]
> Sent: Wednesday, May 05, 2010 8:17 AM
> To: 'freeswitch-users at lists.freeswitch.org'
> Subject: ** Feature Code
>
>
> As you know, dialing **<number> picks up a ringing call (i.e. Directed Call
> Pickup).  However, I noticed yesterday that it will also take a call away
> from another phone even after it is already connected.  Here is the
> scenario:
>
> 1) 5382 calls 5403.
> 2) 5403 answers and two-way talk path.
> 3) 5381 dials **5403
> 4) 5403 loses call.
> 5) 5382 and 5382 two-way talk path.
>
> Is there a way to keep the Directed Call Pickup feature, but prevent taking
> over the call after connect?
>
> Thanks And Best Regards,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <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
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From infos at madovsky.org  Wed May  5 09:37:47 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 5 May 2010 12:37:47 -0400
Subject: [Freeswitch-users] debug DTMF payload
Message-ID: <6D7C014673CA4133908D9691598E6A68@MOBILEE1705>

Hi,

is there any settings in FS that can debug DTMF packets ?
Or any soft that analyze if it's good or bad DTMF packet.

Thanks

F
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From anthony.minessale at gmail.com  Wed May  5 09:40:05 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 5 May 2010 11:40:05 -0500
Subject: [Freeswitch-users] freeSWITCH segfault on openwrt
In-Reply-To: <i2ha44688a1005050627n8739d2bcl280c67dddca79866@mail.gmail.com>
References: <i2ha44688a1005050627n8739d2bcl280c67dddca79866@mail.gmail.com>
Message-ID: <s2m191c3a031005050940h9e74a96cs237802d4f057b0f2@mail.gmail.com>

maybe you are running out of ram.
how many calls do you want on that little toy box?

it's already a package there why are you even compiling it.


On Wed, May 5, 2010 at 8:27 AM, Woody Dickson <woodydickson at gmail.com>wrote:

> Hi,
>
> I manage to get freeSWITCH to compile and run on openwrt, but then
> freeSWITCH keeps crashing when the number of calls reaches 25
> consistently.
>
> Content of the core file returns nothing when opened with gdb.
>
> Does anyone have any clue as to why this is happening and how to fix it?
>
> thanks,
> woody
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <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
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From testeador01 at gmail.com  Wed May  5 10:18:01 2010
From: testeador01 at gmail.com (Milena)
Date: Wed, 5 May 2010 12:18:01 -0500
Subject: [Freeswitch-users] Call for assistance: Spanish sound files,
	translations
In-Reply-To: <i2k87f2f3b91005041315p8375079co4245b4727da0788f@mail.gmail.com>
References: <i2k87f2f3b91005041315p8375079co4245b4727da0788f@mail.gmail.com>
Message-ID: <t2yc29b930b1005051018q85652beg37ec7fed9f997f32@mail.gmail.com>

Hello,

For anyone that reads Michael's email later:

I already completed the file phrase_es.xml, if anyone wants to re-check it
and make it better, it is attached to this message. I also created the
files:

> freeswitch/conf/lang/es.xml

freeswitch/conf/lang/es/vm/sounds.xml

freeswitch/conf/lang/es/dir/sounds.xml

with 2 differences from the english ones (to order the words properly
according to the phrases), check them too if you wish.

Have a nice day

-Milena
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From kward at binarysignal.com  Wed May  5 11:04:39 2010
From: kward at binarysignal.com (Kurt Ward)
Date: Wed, 5 May 2010 11:04:39 -0700
Subject: [Freeswitch-users] External SIP profile
In-Reply-To: <BE1DE3B7-8C40-41E3-B8FE-169A4E9423C3@jerris.com>
References: <56DF79F1-2709-49CC-B5A3-C3A733F2B26E@binarysignal.com>
	<BE1DE3B7-8C40-41E3-B8FE-169A4E9423C3@jerris.com>
Message-ID: <7BA59058-2766-44A5-A7D8-9FFFF0E1AE0D@binarysignal.com>

Thank you for the ideas.  After a lot of trial and error with numerous  
configuration options, I finally found the solution, adding

<param name="extension-in-contact" value="xxxx"/>

to the sofia profile.  Without that, the Mitel 3300 ICP just issues a  
SIP 404 on outgoing calls.  I have added it to the wiki.

Kurt


On May 3, 2010, at 1:20 PM, Michael Jerris wrote:

> yes, roughly the same as a softphone, I think the mitel had some  
> issues with some of our packets, try callerid_in_from
>
> <!--/// replace the INVITE from user with the channel's caller- 
> id ///-->
> <!--/// this is false by default because outbound calls generally  
> need username in the INVITE ///-->
> <param name="caller-id-in-from" value="false"/>
>
> or experiment with other settings from the wiki:
>
> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files
>
> more info on mitel devices:
>
> http://wiki.freeswitch.org/wiki/Interop_List#Mitel_devices
>
> Mike
>
>
> On May 3, 2010, at 12:30 PM, Kurt Ward wrote:
>
>> Maybe this is a bit of an uneducated noob question (or
>> misunderstanding about SIP) but here goes:
>>
>> If I have an external profile set up and registering, does it
>> essentially behave like a soft phone would behave?
>>
>> What I am running into is incoming calls work fine, but outgoing  
>> calls
>> always produce a 404.  If I use a soft phone using the same
>> registration I can make calls with no problem (in both scenarios the
>> calls are to/from 4 digit extensions on a Mitel PBX).
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From freeswitch at bsdjournal.net  Wed May  5 11:40:03 2010
From: freeswitch at bsdjournal.net (Bryan Vyhmeister)
Date: Wed, 5 May 2010 14:40:03 -0400
Subject: [Freeswitch-users] Sangoma prid and multiple PRIs with
	FreeSWITCH
In-Reply-To: <x2y7e9692c31005050801ud0a264f6u709549a8790e10b0@mail.gmail.com>
References: <x2y7e9692c31005050801ud0a264f6u709549a8790e10b0@mail.gmail.com>
Message-ID: <i2i7e9692c31005051140q25335e11seae30abf42c03d89@mail.gmail.com>

I just spoke with Sangoma tech support. I am posting a workaround solution
here in case someone else runs into the same problem. As long as both spans
use the same switch type, FreeSWITCH can see both as the same span. In
/etc/wanpipe/smg_pri.conf, you need to define a separate group for each
span.

;AFT-A102 on port 1
signalling=pri_cpe
switchtype=national
group=1
spans=1

;AFT-A102 on port 2
signalling=pri_cpe
switchtype=national
group=2
spans=2

In openzap.conf.xml you only need one span:

<configuration name="openzap.conf" description="OpenZAP Configuration">
  <settings>
    <param name="debug" value="0"/>
  </settings>
  <boost_spans>
<span id="1">
        <param name="node" value="cpe"/>
        <param name="switch" value="ni2"/>
<param name="dialplan" value="XML"/>
<param name="context" value="public"/>
</span>
  </boost_spans>
   <analog_spans>
  </analog_spans>
 </configuration>

And the same in openzap.conf however you need to define the channels of both
physical PRI circuits.

[span wanpipe 1]
name => 1
trunk_type =>t1
b-channel => 1:1-23
b-channel => 2:1-23

When you dial, instead of using openzap/1 or openzap/2 you have to use
groups. This would be the first PRI dial string:

<action application="bridge" data="openzap/1/a/$1 at g1"/>

This would be the second PRI dial string:

<action application="bridge" data="openzap/1/a/$1 at g2"/>

This works as expected. Apparently there will be a new way of interacting
with smg_prid coming out at some point that will address these issues and
allow multiple spans and switch types in the openzap configs. Thanks to
Sangoma for the great hardware and drivers and the FreeSWITCH project for a
fantastic product!

Bryan


On Wed, May 5, 2010 at 11:01 AM, Bryan Vyhmeister <freeswitch at bsdjournal.net
> wrote:

> I setup a FreeSWITCH pbx with a Sangoma A102D card, the latest FreeSWITCH
> as of yesterday, and Wanpipe 3.5.11. I have found that I cannot use both PRI
> ports with Sangoma's smg_prid. The first one always works great but the
> second one never works. I get the following error:
>
> 2010-05-05 09:42:02.573743 [CRIT] ozmod_sangoma_boost.c:257 SPAN is not
> online.
> 2010-05-05 09:42:02.573743 [ERR] switch_ivr_originate.c:2480 Cannot create
> outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
>
> When I start FreeSWITCH up I get this error on the console right after all
> the channels are activated.
>
> 2010-05-05 07:25:59.414444 [ERR] ozmod_sangoma_boost.c:1534 Error: Opening
> MCON Socket [-1] Address already in use
>
> The configs are below:
>
> openzap.conf
> [span wanpipe 1]
> name => 1
> trunk_type =>t1
> b-channel => 1:1-23
> [span wanpipe 2]
> name => 2
> trunk_type =>t1
> b-channel => 2:1-23
>
> openzap.conf.xml
> <configuration name="openzap.conf" description="OpenZAP Configuration">
>   <settings>
>     <param name="debug" value="0"/>
>   </settings>
>   <boost_spans>
> <span id="1">
>         <param name="node" value="cpe"/>
>         <param name="switch" value="ni2"/>
> <param name="dialplan" value="XML"/>
>  <param name="context" value="public"/>
> </span>
>  <span id="2">
>         <param name="node" value="cpe"/>
>         <param name="switch" value="ni2"/>
>  <param name="dialplan" value="XML"/>
> <param name="context" value="public"/>
>   </boost_spans>
>   <analog_spans>
>   </analog_spans>
> </configuration>
>
> I'm dialing with this dial string.
>
>         <action application="bridge"
> data="{origination_caller_id_number=${outbound_caller_id_number}}openzap/2/a/$1"/>
>
> I am certain the rest of this config works fine because the other PRI works
> great. I have also attempted to use FreeSWITCH without Sangoma's smg_prid by
> specifying a d-channel in openzap.conf. That seems to work but calls tend to
> drop frequently. I would rather just use Sangoma's smg_prid. I also
> attempted a hybrid config where FreeSWITCH worked without smg_prid on the
> first PRI and the second PRI was handled by smg_prid. That config worked
> except for the dropped calls that happen with using FreeSWITCH to manage the
> d-channel.
>
> My question is how can I get two PRI's working with Sangoma's smg_prid?
> Thank you.
>
> Bryan
>
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From msc at freeswitch.org  Wed May  5 12:26:37 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 May 2010 12:26:37 -0700
Subject: [Freeswitch-users] How to make scheduled calls ?
In-Reply-To: <o2rd9f1e4661005050823q704bf7feje4a19d216eef98fe@mail.gmail.com>
References: <o2rd9f1e4661005050823q704bf7feje4a19d216eef98fe@mail.gmail.com>
Message-ID: <j2m87f2f3b91005051226sdde19099y3633e51d0e00f31f@mail.gmail.com>

Look at sched_api and originate commands on the wiki.
-MC

On Wed, May 5, 2010 at 8:23 AM, hung nguyen <hungngm1987 at gmail.com> wrote:

> Hi list.
>
> I want config FS to make automatic calls follow a fixed schedule.
> Example: In fixed time in the future, FS will call a fixed user and play a
> fixed music file for this user.
>
> I don't want to use crontab to do this because i want to deploy large
> schedules.
>
> Thank to suggestions.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed May  5 12:31:32 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 May 2010 12:31:32 -0700
Subject: [Freeswitch-users] debug DTMF payload
In-Reply-To: <6D7C014673CA4133908D9691598E6A68@MOBILEE1705>
References: <6D7C014673CA4133908D9691598E6A68@MOBILEE1705>
Message-ID: <v2j87f2f3b91005051231kca2fbafam2cd3445976445fda@mail.gmail.com>

tcpdump + Wireshark will let you see DTMFs out of band. If you need to see
them inband you'll need some sort of software to analyze a recording of the
call. IIRC, Audacity has some stuff in it for that.
-MC

On Wed, May 5, 2010 at 9:37 AM, Madovsky <infos at madovsky.org> wrote:

>  Hi,
>
> is there any settings in FS that can debug DTMF packets ?
> Or any soft that analyze if it's good or bad DTMF packet.
>
> Thanks
>
> F
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed May  5 12:33:49 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 May 2010 12:33:49 -0700
Subject: [Freeswitch-users] Sangoma prid and multiple PRIs with
	FreeSWITCH
In-Reply-To: <i2i7e9692c31005051140q25335e11seae30abf42c03d89@mail.gmail.com>
References: <x2y7e9692c31005050801ud0a264f6u709549a8790e10b0@mail.gmail.com>
	<i2i7e9692c31005051140q25335e11seae30abf42c03d89@mail.gmail.com>
Message-ID: <y2p87f2f3b91005051233vbc4bd779n3c379f0c748d6a36@mail.gmail.com>

I believe we started a boost section on the openzap wiki page. If not, would
you mind adding these config examples? It would be very helpful for the next
poor soul coming along trying to work on this.

Thanks,
MC

On Wed, May 5, 2010 at 11:40 AM, Bryan Vyhmeister <freeswitch at bsdjournal.net
> wrote:

> I just spoke with Sangoma tech support. I am posting a workaround solution
> here in case someone else runs into the same problem. As long as both spans
> use the same switch type, FreeSWITCH can see both as the same span. In
> /etc/wanpipe/smg_pri.conf, you need to define a separate group for each
> span.
>
> ;AFT-A102 on port 1
> signalling=pri_cpe
> switchtype=national
> group=1
> spans=1
>
> ;AFT-A102 on port 2
> signalling=pri_cpe
> switchtype=national
> group=2
> spans=2
>
> In openzap.conf.xml you only need one span:
>
> <configuration name="openzap.conf" description="OpenZAP Configuration">
>   <settings>
>     <param name="debug" value="0"/>
>   </settings>
>   <boost_spans>
> <span id="1">
>         <param name="node" value="cpe"/>
>         <param name="switch" value="ni2"/>
> <param name="dialplan" value="XML"/>
>  <param name="context" value="public"/>
> </span>
>   </boost_spans>
>    <analog_spans>
>   </analog_spans>
>  </configuration>
>
> And the same in openzap.conf however you need to define the channels of
> both physical PRI circuits.
>
> [span wanpipe 1]
> name => 1
> trunk_type =>t1
> b-channel => 1:1-23
> b-channel => 2:1-23
>
> When you dial, instead of using openzap/1 or openzap/2 you have to use
> groups. This would be the first PRI dial string:
>
> <action application="bridge" data="openzap/1/a/$1 at g1"/>
>
> This would be the second PRI dial string:
>
> <action application="bridge" data="openzap/1/a/$1 at g2"/>
>
> This works as expected. Apparently there will be a new way of interacting
> with smg_prid coming out at some point that will address these issues and
> allow multiple spans and switch types in the openzap configs. Thanks to
> Sangoma for the great hardware and drivers and the FreeSWITCH project for a
> fantastic product!
>
> Bryan
>
>
> On Wed, May 5, 2010 at 11:01 AM, Bryan Vyhmeister <
> freeswitch at bsdjournal.net> wrote:
>
>> I setup a FreeSWITCH pbx with a Sangoma A102D card, the latest FreeSWITCH
>> as of yesterday, and Wanpipe 3.5.11. I have found that I cannot use both PRI
>> ports with Sangoma's smg_prid. The first one always works great but the
>> second one never works. I get the following error:
>>
>> 2010-05-05 09:42:02.573743 [CRIT] ozmod_sangoma_boost.c:257 SPAN is not
>> online.
>> 2010-05-05 09:42:02.573743 [ERR] switch_ivr_originate.c:2480 Cannot create
>> outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
>>
>> When I start FreeSWITCH up I get this error on the console right after all
>> the channels are activated.
>>
>> 2010-05-05 07:25:59.414444 [ERR] ozmod_sangoma_boost.c:1534 Error: Opening
>> MCON Socket [-1] Address already in use
>>
>> The configs are below:
>>
>> openzap.conf
>> [span wanpipe 1]
>> name => 1
>> trunk_type =>t1
>> b-channel => 1:1-23
>> [span wanpipe 2]
>> name => 2
>> trunk_type =>t1
>> b-channel => 2:1-23
>>
>> openzap.conf.xml
>> <configuration name="openzap.conf" description="OpenZAP Configuration">
>>   <settings>
>>     <param name="debug" value="0"/>
>>   </settings>
>>   <boost_spans>
>> <span id="1">
>>         <param name="node" value="cpe"/>
>>         <param name="switch" value="ni2"/>
>> <param name="dialplan" value="XML"/>
>>  <param name="context" value="public"/>
>> </span>
>>  <span id="2">
>>         <param name="node" value="cpe"/>
>>         <param name="switch" value="ni2"/>
>>  <param name="dialplan" value="XML"/>
>> <param name="context" value="public"/>
>>   </boost_spans>
>>   <analog_spans>
>>   </analog_spans>
>> </configuration>
>>
>> I'm dialing with this dial string.
>>
>>         <action application="bridge"
>> data="{origination_caller_id_number=${outbound_caller_id_number}}openzap/2/a/$1"/>
>>
>> I am certain the rest of this config works fine because the other PRI
>> works great. I have also attempted to use FreeSWITCH without Sangoma's
>> smg_prid by specifying a d-channel in openzap.conf. That seems to work but
>> calls tend to drop frequently. I would rather just use Sangoma's smg_prid. I
>> also attempted a hybrid config where FreeSWITCH worked without smg_prid on
>> the first PRI and the second PRI was handled by smg_prid. That config worked
>> except for the dropped calls that happen with using FreeSWITCH to manage the
>> d-channel.
>>
>> My question is how can I get two PRI's working with Sangoma's smg_prid?
>> Thank you.
>>
>> Bryan
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Wed May  5 12:35:19 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 5 May 2010 14:35:19 -0500
Subject: [Freeswitch-users] How to make scheduled calls ?
In-Reply-To: <j2m87f2f3b91005051226sdde19099y3633e51d0e00f31f@mail.gmail.com>
References: <o2rd9f1e4661005050823q704bf7feje4a19d216eef98fe@mail.gmail.com>
	<j2m87f2f3b91005051226sdde19099y3633e51d0e00f31f@mail.gmail.com>
Message-ID: <A8EFD586-B965-4DD7-8047-FF73F37216BA@freeswitch.org>

man cron

/b

On May 5, 2010, at 2:26 PM, Michael Collins wrote:

> Look at sched_api and originate commands on the wiki.
> -MC



From kenfulmer at icstechnologysolutions.com  Wed May  5 12:41:17 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Wed, 5 May 2010 14:41:17 -0500
Subject: [Freeswitch-users] Sorting through a list of conditions
Message-ID: <017a01caec8a$ee74bb70$cb5e3250$@com>

We'd like to match on either Caller ID or ANI for outbound faxes and lock
them to PCMU. 

 

Here's what we've been using:

 

<extension name="FAX-LCL-7D">

   <condition field="caller_id_number" expression="^XXXXXXXXXX$"/>

   <condition field="destination_number" expression="^9(\d{7})$">

       <action application="export" data="sip_append_audio_sdp=a=fmtp:18
annexb=no"/>

       <action application="export"
data="nolocal:absolute_codec_string=PCMU"/>

       <action application="set" data="hangup_after_bridge=true"/>

       <action application="bridge" data="sofia/external/$0 at X.X.X.X"/>

       <action application="set" data="instant_ringback=true"/>

       <action application="set" data="ringback=${us-ring}"/>

   </condition>

</extension>

 

If we use a list and break=on-false, searching stops after the first
non-match. If we use break=on-true, it stops after the first match, and we
get no digits to send upstream. If we use break=never, it applies the digit
match to all numbers, regardless of Caller ID. I realize this is all normal
behavior. 

 

Is there a way to sort through a list of Caller ID's, match one, and then
match the outbound digits? We're trying to avoid having to create a
dial-plan entry for every Caller ID. 

 

Here's what it *might* look like with multiple internal fax lines:

 

<extension name="FAX-LCL-7D">

   <condition field="caller_id_number" expression="^XXXXXXXXXX$"/>

   <condition field="caller_id_number" expression="^XXXXXXXXXX$"/>

   <condition field="caller_id_number" expression="^XXXXXXXXXX$"/>

   <condition field="caller_id_number" expression="^XXXXXXXXXX$"/>

   <condition field="caller_id_number" expression="^XXXXXXXXXX$"/>

   <condition field="caller_id_number" expression="^XXXXXXXXXX$"/>

   <condition field="caller_id_number" expression="^XXXXXXXXXX$"/>

   <condition field="caller_id_number" expression="^XXXXXXXXXX$"/>

   <condition field="destination_number" expression="^9(\d{7})$">

       <action application="export" data="sip_append_audio_sdp=a=fmtp:18
annexb=no"/>

       <action application="export"
data="nolocal:absolute_codec_string=PCMU"/>

       <action application="set" data="hangup_after_bridge=true"/>

       <action application="bridge" data="sofia/external/$0 at X.X.X.X"/>

       <action application="set" data="instant_ringback=true"/>

       <action application="set" data="ringback=${us-ring}"/>

   </condition>

</extension>

 

 

Thanks,

 

Ken Fulmer

 

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From david.ponzone at gmail.com  Wed May  5 12:57:20 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Wed, 5 May 2010 21:57:20 +0200
Subject: [Freeswitch-users] Sorting through a list of conditions
In-Reply-To: <017a01caec8a$ee74bb70$cb5e3250$@com>
References: <017a01caec8a$ee74bb70$cb5e3250$@com>
Message-ID: <BE3F6D4F-9486-43F3-ACAF-70C84CB001FF@gmail.com>

Ken,

ever heard of regexp ? :)

In this case, I think you need to use one, or to add one extension per  
fax line, which can be more readable and managable.
Other solution would be to use an external DB with XML CURL.

The regexp solution would be:

    <condition field="caller_id_number" expression="^XXXXXXXXXX| 
XXXXXXXXXX|XXXXXXXXXX|XXXXXXXXXX|XXXXXXXXXX$"/>

Another smart (I think) way to do that would be to create and dedicate  
a new SIP Profile on a specific IP or port to fax handling (of course,  
that's only possible if you use a dedicated SIP account for fax).

I think the ability to split FreeSWITCH into multiple SIP Profiles  
(and so multiple threads) is something that should not be underrated.
It's really easier to manage and you dont break everything when you  
test some new parameters on the profile.
Most serious SIP devices should allow to register to a specific port,  
so you can use something else than 5060 (Ok I admit some shitty gear  
don't allow that, but most of the ATAs I know are ok).


David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 05/05/2010 ? 21:41, Ken Fulmer a ?crit :

> We?d like to match on either Caller ID or ANI for outbound faxes  
> and lock them to PCMU.
>
> Here?s what we?ve been using:
>
> <extension name="FAX-LCL-7D">
>    <condition field="caller_id_number" expression="^XXXXXXXXXX$"/>
>    <condition field="destination_number" expression="^9(\d{7})$">
>        <action application="export"  
> data="sip_append_audio_sdp=a=fmtp:18 annexb=no"/>
>        <action application="export"  
> data="nolocal:absolute_codec_string=PCMU"/>
>        <action application="set" data="hangup_after_bridge=true"/>
>        <action application="bridge" data="sofia/external/$0 at X.X.X.X"/>
>        <action application="set" data="instant_ringback=true"/>
>        <action application="set" data="ringback=${us-ring}"/>
>    </condition>
> </extension>
>
> If we use a list and break=on-false, searching stops after the first  
> non-match. If we use break=on-true, it stops after the first match,  
> and we get no digits to send upstream. If we use break=never, it  
> applies the digit match to all numbers, regardless of Caller ID. I  
> realize this is all normal behavior.
>
> Is there a way to sort through a list of Caller ID?s, match one,  
> and then match the outbound digits? We?re trying to avoid having to  
> create a dial-plan entry for every Caller ID.
>
> Here?s what it *might* look like with multiple internal fax lines:
>
> <extension name="FAX-LCL-7D">
>    <condition field="caller_id_number" expression="^XXXXXXXXXX$"/>
>    <condition field="caller_id_number" expression="^XXXXXXXXXX$"/>
>    <condition field="caller_id_number" expression="^XXXXXXXXXX$"/>
>    <condition field="caller_id_number" expression="^XXXXXXXXXX$"/>
>    <condition field="caller_id_number" expression="^XXXXXXXXXX$"/>
>    <condition field="caller_id_number" expression="^XXXXXXXXXX$"/>
>    <condition field="caller_id_number" expression="^XXXXXXXXXX$"/>
>    <condition field="caller_id_number" expression="^XXXXXXXXXX$"/>
>    <condition field="destination_number" expression="^9(\d{7})$">
>        <action application="export"  
> data="sip_append_audio_sdp=a=fmtp:18 annexb=no"/>
>        <action application="export"  
> data="nolocal:absolute_codec_string=PCMU"/>
>        <action application="set" data="hangup_after_bridge=true"/>
>        <action application="bridge" data="sofia/external/$0 at X.X.X.X"/>
>        <action application="set" data="instant_ringback=true"/>
>        <action application="set" data="ringback=${us-ring}"/>
>    </condition>
> </extension>
>
>
> Thanks,
>
> Ken Fulmer
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From testeador01 at gmail.com  Wed May  5 13:00:25 2010
From: testeador01 at gmail.com (Milena)
Date: Wed, 5 May 2010 15:00:25 -0500
Subject: [Freeswitch-users] Sorting through a list of conditions
In-Reply-To: <017a01caec8a$ee74bb70$cb5e3250$@com>
References: <017a01caec8a$ee74bb70$cb5e3250$@com>
Message-ID: <o2nc29b930b1005051300s10da268q681bb273a1f61776@mail.gmail.com>

Hello,

I'm not quite sure I understood what you're asking, but i think what you
want to do is:

<extension name="FAX-LCL-7D">
   <condition field="caller_id_number"
expression="^((XXXXXXXXXX)|(XXXXXXXXXY)|(XXXXXXXXXZ))$"/>
   <condition field="destination_number" expression="^9(\d{7})$">
     ( ... )

Is that it?

-Milena

2010/5/5 Ken Fulmer <kenfulmer at icstechnologysolutions.com>

>  We?d like to match on either Caller ID or ANI for outbound faxes and lock
> them to PCMU.
>
>
>
> Here?s what we?ve been using:
>
>
>
> <extension name="FAX-LCL-7D">
>
>    <condition field="caller_id_number" expression="^XXXXXXXXXX$"/>
>
>    <condition field="destination_number" expression="^9(\d{7})$">
>
>        <action application="export" data="sip_append_audio_sdp=a=fmtp:18
> annexb=no"/>
>
>        <action application="export"
> data="nolocal:absolute_codec_string=PCMU"/>
>
>        <action application="set" data="hangup_after_bridge=true"/>
>
>        <action application="bridge" data="sofia/external/$0 at X.X.X.X"/>
>
>        <action application="set" data="instant_ringback=true"/>
>
>        <action application="set" data="ringback=${us-ring}"/>
>
>    </condition>
>
> </extension>
>
>
>
> If we use a list and break=on-false, searching stops after the first
> non-match. If we use break=on-true, it stops after the first match, and we
> get no digits to send upstream. If we use break=never, it applies the digit
> match to all numbers, regardless of Caller ID. I realize this is all normal
> behavior.
>
>
>
> Is there a way to sort through a list of Caller ID?s, match one, and then
> match the outbound digits? We?re trying to avoid having to create a
> dial-plan entry for every Caller ID.
>
>
>
> Here?s what it *might* look like with multiple internal fax lines:
>
>
>
> <extension name="FAX-LCL-7D">
>
>    <condition field="caller_id_number" expression="^XXXXXXXXXX$"/>
>
>    <condition field="caller_id_number" expression="^XXXXXXXXXX$"/>
>
>    <condition field="caller_id_number" expression="^XXXXXXXXXX$"/>
>
>    <condition field="caller_id_number" expression="^XXXXXXXXXX$"/>
>
>    <condition field="caller_id_number" expression="^XXXXXXXXXX$"/>
>
>    <condition field="caller_id_number" expression="^XXXXXXXXXX$"/>
>
>    <condition field="caller_id_number" expression="^XXXXXXXXXX$"/>
>
>    <condition field="caller_id_number" expression="^XXXXXXXXXX$"/>
>
>    <condition field="destination_number" expression="^9(\d{7})$">
>
>        <action application="export" data="sip_append_audio_sdp=a=fmtp:18
> annexb=no"/>
>
>        <action application="export"
> data="nolocal:absolute_codec_string=PCMU"/>
>
>        <action application="set" data="hangup_after_bridge=true"/>
>
>        <action application="bridge" data="sofia/external/$0 at X.X.X.X"/>
>
>        <action application="set" data="instant_ringback=true"/>
>
>        <action application="set" data="ringback=${us-ring}"/>
>
>    </condition>
>
> </extension>
>
>
>
>
>
> Thanks,
>
>
>
> Ken Fulmer
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From vfclists at googlemail.com  Wed May  5 13:27:03 2010
From: vfclists at googlemail.com (Frank Church)
Date: Wed, 5 May 2010 21:27:03 +0100
Subject: [Freeswitch-users] Are Freeswitch epochs strictly unixtime?
Message-ID: <h2v7abab2411005051327macb01906w2744aff4987bef4f@mail.gmail.com>

After converting an epoch to a Microsoft date time value I realized that the
result was an hour behind. I am in British Summer Time now.

I just want to confirm that all epoch values in Freeswitch are independent
of time zone.

-- 
Frank Church

=======================
http://devblog.brahmancreations.com
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From faxport at hotmail.com  Wed May  5 13:38:08 2010
From: faxport at hotmail.com (C Sarti)
Date: Wed, 5 May 2010 20:38:08 +0000
Subject: [Freeswitch-users] Freeswitch - Polycom IP 300
Message-ID: <SNT135-w42907186A87E973DBC1CC6A9F40@phx.gbl>


Hi I dont know if I should post here for help.. Im new with FS.

 

I have installed FS in win2003 , with softphone so far is working fine,

 

I am trying to use couple of Polycom 300 I had with callcentric  (no luck :( )

 

at the FS console I have anable siptrace and got the following error:

 
SIP/2.0 401 Unauthorized.
  via: sip/2.0/udp 192.168.0.2;branch=z9hG4bke36814c786B88EF;received= ......

honestly, from here on Im lost :(  any idea what to do?thank you in advanced.regardsCarlos  

  		 	   		  
_________________________________________________________________
Hotmail is redefining busy with tools for the New Busy. Get more from your inbox.
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From brian at freeswitch.org  Wed May  5 13:45:38 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 5 May 2010 15:45:38 -0500
Subject: [Freeswitch-users] Freeswitch - Polycom IP 300
In-Reply-To: <SNT135-w42907186A87E973DBC1CC6A9F40@phx.gbl>
References: <SNT135-w42907186A87E973DBC1CC6A9F40@phx.gbl>
Message-ID: <2ED36688-94A9-4D4F-8993-AEE5AE39D84F@freeswitch.org>

You have nat involved don't you?

/b

On May 5, 2010, at 3:38 PM, C Sarti wrote:

> honestly, from here on Im lost :(  any idea what to do?
> thank you in advanced.
> 

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From kenfulmer at icstechnologysolutions.com  Wed May  5 13:47:04 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Wed, 5 May 2010 15:47:04 -0500
Subject: [Freeswitch-users] Sorting through a list of conditions
In-Reply-To: <BE3F6D4F-9486-43F3-ACAF-70C84CB001FF@gmail.com>
References: <017a01caec8a$ee74bb70$cb5e3250$@com>
	<BE3F6D4F-9486-43F3-ACAF-70C84CB001FF@gmail.com>
Message-ID: <01b101caec94$1f009fd0$5d01df70$@com>

That is exactly what we needed. I?m still a little new to matching with regular expressions.

 

Thanks!

 

Ken

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Ponzone
Sent: Wednesday, May 05, 2010 2:57 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Sorting through a list of conditions

 

Ken,

 

ever heard of regexp ? :)

 

In this case, I think you need to use one, or to add one extension per fax line, which can be more readable and managable.

Other solution would be to use an external DB with XML CURL.

 

The regexp solution would be:

 

   <condition field="caller_id_number" expression="^XXXXXXXXXX|XXXXXXXXXX|XXXXXXXXXX|XXXXXXXXXX|XXXXXXXXXX$"/>

 

Another smart (I think) way to do that would be to create and dedicate a new SIP Profile on a specific IP or port to fax handling (of course, that's only possible if you use a dedicated SIP account for fax).

 

I think the ability to split FreeSWITCH into multiple SIP Profiles (and so multiple threads) is something that should not be underrated.

It's really easier to manage and you dont break everything when you test some new parameters on the profile.

Most serious SIP devices should allow to register to a specific port, so you can use something else than 5060 (Ok I admit some shitty gear don't allow that, but most of the ATAs I know are ok).

 

 

David Ponzone  Direction Technique

email: david.ponzone at ipeva.fr

tel:      01 74 03 18 97

gsm:   06 66 98 76 34

 

Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>   -   www.ipeva-studio.com <BLOCKED::http://www.ipeva-studio.com/> 

 

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.

 

 

 

 

Le 05/05/2010 ? 21:41, Ken Fulmer a ?crit :





We?d like to match on either Caller ID or ANI for outbound faxes and lock them to PCMU.

 

Here?s what we?ve been using:

 

<extension name="FAX-LCL-7D">

   <condition field="caller_id_number" expression="^XXXXXXXXXX$"/>

   <condition field="destination_number" expression="^9(\d{7})$">

       <action application="export" data="sip_append_audio_sdp=a=fmtp:18 annexb=no"/>

       <action application="export" data="nolocal:absolute_codec_string=PCMU"/>

       <action application="set" data="hangup_after_bridge=true"/>

       <action application="bridge" data="sofia/external/$0 at X.X.X.X"/>

       <action application="set" data="instant_ringback=true"/>

       <action application="set" data="ringback=${us-ring}"/>

   </condition>

</extension>

 

If we use a list and break=on-false, searching stops after the first non-match. If we use break=on-true, it stops after the first match, and we get no digits to send upstream. If we use break=never, it applies the digit match to all numbers, regardless of Caller ID. I realize this is all normal behavior.

 

Is there a way to sort through a list of Caller ID?s, match one, and then match the outbound digits? We?re trying to avoid having to create a dial-plan entry for every Caller ID.

 

Here?s what it *might* look like with multiple internal fax lines:

 

<extension name="FAX-LCL-7D">

   <condition field="caller_id_number" expression="^XXXXXXXXXX$"/>

   <condition field="caller_id_number" expression="^XXXXXXXXXX$"/>

   <condition field="caller_id_number" expression="^XXXXXXXXXX$"/>

   <condition field="caller_id_number" expression="^XXXXXXXXXX$"/>

   <condition field="caller_id_number" expression="^XXXXXXXXXX$"/>

   <condition field="caller_id_number" expression="^XXXXXXXXXX$"/>

   <condition field="caller_id_number" expression="^XXXXXXXXXX$"/>

   <condition field="caller_id_number" expression="^XXXXXXXXXX$"/>

   <condition field="destination_number" expression="^9(\d{7})$">

       <action application="export" data="sip_append_audio_sdp=a=fmtp:18 annexb=no"/>

       <action application="export" data="nolocal:absolute_codec_string=PCMU"/>

       <action application="set" data="hangup_after_bridge=true"/>

       <action application="bridge" data="sofia/external/$0 at X.X.X.X"/>

       <action application="set" data="instant_ringback=true"/>

       <action application="set" data="ringback=${us-ring}"/>

   </condition>

</extension>

 

 

Thanks,

 

Ken Fulmer

 

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

 

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From msc at freeswitch.org  Wed May  5 13:55:18 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 May 2010 13:55:18 -0700
Subject: [Freeswitch-users] Sorting through a list of conditions
In-Reply-To: <01b101caec94$1f009fd0$5d01df70$@com>
References: <017a01caec8a$ee74bb70$cb5e3250$@com>
	<BE3F6D4F-9486-43F3-ACAF-70C84CB001FF@gmail.com>
	<01b101caec94$1f009fd0$5d01df70$@com>
Message-ID: <w2x87f2f3b91005051355j7c5c775dld43c79c6bf9d61cb@mail.gmail.com>

On Wed, May 5, 2010 at 1:47 PM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  That is exactly what we needed. I?m still a little new to matching with
> regular expressions.
>
You'll be interested in Chapter 5 of the upcoming FreeSWITCH book... :D
-MC
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From anthony.minessale at gmail.com  Wed May  5 14:14:28 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 5 May 2010 16:14:28 -0500
Subject: [Freeswitch-users] Are Freeswitch epochs strictly unixtime?
In-Reply-To: <h2v7abab2411005051327macb01906w2744aff4987bef4f@mail.gmail.com>
References: <h2v7abab2411005051327macb01906w2744aff4987bef4f@mail.gmail.com>
Message-ID: <t2u191c3a031005051414scb7f032bp65bbeaf860874ea1@mail.gmail.com>

they would be if you set the box to UTC timezone
they use the local time of the box which on a server is typically wise to be
greenwich

On Wed, May 5, 2010 at 3:27 PM, Frank Church <vfclists at googlemail.com>wrote:

> After converting an epoch to a Microsoft date time value I realized that
> the result was an hour behind. I am in British Summer Time now.
>
> I just want to confirm that all epoch values in Freeswitch are independent
> of time zone.
>
> --
> Frank Church
>
> =======================
> http://devblog.brahmancreations.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <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
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From Prometheus001 at gmx.net  Wed May  5 16:13:06 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Thu, 06 May 2010 01:13:06 +0200
Subject: [Freeswitch-users] mod_com_g729 DECODER CREATE FAILED
In-Reply-To: <AANLkTimPZkIXxd0PIoqxyM9L5KNU7__F2TYiMLmCqK7h@mail.gmail.com>
References: <4BD85C5F.3000302@gmx.net>	<y2u191c3a031004281324j26173b32nc89a6b0f9716871d@mail.gmail.com>	<4BDABDA9.5020707@gmx.net>	<AANLkTilHq4lZ_qt9-XjAVbvn7Dlur9u-7a28MERi_Bao@mail.gmail.com>
	<AANLkTimPZkIXxd0PIoqxyM9L5KNU7__F2TYiMLmCqK7h@mail.gmail.com>
Message-ID: <4BE1FB82.3040708@gmx.net>

Hello Anthony,

thanks, I received your mail. I should use a different email adreess in
the future. My email provider sometimes sends these strange
notifications as you have seen. eMails from this mailing lists however
work. Every attempt to stop this behaviour failed, as the email provider
has no clue how to solve this. Sorry for this inconvenience.

Concerning the G.729 problem, this is not 100% clear to me.
When I connect 2 phones with G.729 without recording, I can see that no
license is used, as the call is in bypass mode:
  Permitted G.729AB channels: 2
  Encoders in use: 0
  Decoders in use: 0

So when I try to record this call, I will need 3 licenses? 1 for each
call leg and one for the recording?

Best regards
Peter



From brian at freeswitch.org  Wed May  5 16:19:04 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 5 May 2010 18:19:04 -0500
Subject: [Freeswitch-users] mod_com_g729 DECODER CREATE FAILED
In-Reply-To: <4BE1FB82.3040708@gmx.net>
References: <4BD85C5F.3000302@gmx.net>	<y2u191c3a031004281324j26173b32nc89a6b0f9716871d@mail.gmail.com>	<4BDABDA9.5020707@gmx.net>	<AANLkTilHq4lZ_qt9-XjAVbvn7Dlur9u-7a28MERi_Bao@mail.gmail.com>
	<AANLkTimPZkIXxd0PIoqxyM9L5KNU7__F2TYiMLmCqK7h@mail.gmail.com>
	<4BE1FB82.3040708@gmx.net>
Message-ID: <BDE760B2-81DA-4F99-A718-264F9B348851@freeswitch.org>

Well if you have a g729 call up and its G729 passthru encoder/decoders will not be allocated till needed.  If you record the call you'll require two decoders to record the call since it has to decode both sides and mux the call.  I did test calls that were G729 end to end and no coders got allocated.  The allocation happened when recording started.  Also if you do fake ringback you'll allocate coders too.

/b

On May 5, 2010, at 6:13 PM, Peter P GMX wrote:

> Hello Anthony,
> 
> thanks, I received your mail. I should use a different email adreess in
> the future. My email provider sometimes sends these strange
> notifications as you have seen. eMails from this mailing lists however
> work. Every attempt to stop this behaviour failed, as the email provider
> has no clue how to solve this. Sorry for this inconvenience.
> 
> Concerning the G.729 problem, this is not 100% clear to me.
> When I connect 2 phones with G.729 without recording, I can see that no
> license is used, as the call is in bypass mode:
>  Permitted G.729AB channels: 2
>  Encoders in use: 0
>  Decoders in use: 0
> 
> So when I try to record this call, I will need 3 licenses? 1 for each
> call leg and one for the recording?
> 
> Best regards
> Peter



From anthony.minessale at gmail.com  Wed May  5 16:32:52 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 5 May 2010 18:32:52 -0500
Subject: [Freeswitch-users] mod_com_g729 DECODER CREATE FAILED
In-Reply-To: <4BE1FB82.3040708@gmx.net>
References: <4BD85C5F.3000302@gmx.net>
	<y2u191c3a031004281324j26173b32nc89a6b0f9716871d@mail.gmail.com>
	<4BDABDA9.5020707@gmx.net>
	<AANLkTilHq4lZ_qt9-XjAVbvn7Dlur9u-7a28MERi_Bao@mail.gmail.com>
	<AANLkTimPZkIXxd0PIoqxyM9L5KNU7__F2TYiMLmCqK7h@mail.gmail.com>
	<4BE1FB82.3040708@gmx.net>
Message-ID: <s2g191c3a031005051632g4116eeb9nf2e1aa35f9be0272@mail.gmail.com>

if the call is in passthru it will use 0 licenses
if you begin to record it will use 2 decoder until you stop recording

if the call is transcoding to g729 to ulaw and you are careful to record on
the ulaw leg it will cost no extra licenses other than the one already in
use  otherwise it will cost 1 if you do it on the 729 side.


On Wed, May 5, 2010 at 6:13 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> Hello Anthony,
>
> thanks, I received your mail. I should use a different email adreess in
> the future. My email provider sometimes sends these strange
> notifications as you have seen. eMails from this mailing lists however
> work. Every attempt to stop this behaviour failed, as the email provider
> has no clue how to solve this. Sorry for this inconvenience.
>
> Concerning the G.729 problem, this is not 100% clear to me.
> When I connect 2 phones with G.729 without recording, I can see that no
> license is used, as the call is in bypass mode:
>   Permitted G.729AB channels: 2
>  Encoders in use: 0
>  Decoders in use: 0
>
> So when I try to record this call, I will need 3 licenses? 1 for each
> call leg and one for the recording?
>
> Best regards
> Peter
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
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
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From woodydickson at gmail.com  Wed May  5 16:58:47 2010
From: woodydickson at gmail.com (Woody Dickson)
Date: Thu, 6 May 2010 07:58:47 +0800
Subject: [Freeswitch-users] freeSWITCH segfault on openwrt
In-Reply-To: <s2m191c3a031005050940h9e74a96cs237802d4f057b0f2@mail.gmail.com>
References: <i2ha44688a1005050627n8739d2bcl280c67dddca79866@mail.gmail.com>
	<s2m191c3a031005050940h9e74a96cs237802d4f057b0f2@mail.gmail.com>
Message-ID: <o2ya44688a1005051658q204f4507w2e9b118e4289e9ab@mail.gmail.com>

Hi,

I have 2 G of RAM in that box and with 25 calls, there are still
plenty of memory left.  Is there anyway I can debug this issue?

Are you referring to the astlinux freeswitch package?  I can't seem to
find it from the astlinux site.

I found a link from some old threads pointing to the astlinux download
page, but the iso image is not there.

Can someone please point me to the correct download page?

thanks,
woody


On Thu, May 6, 2010 at 12:40 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> maybe you are running out of ram.
> how many calls do you want on that little toy box?
> it's already a package there why are you even compiling it.
>


From anthony.minessale at gmail.com  Wed May  5 17:06:38 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 5 May 2010 19:06:38 -0500
Subject: [Freeswitch-users] freeSWITCH segfault on openwrt
In-Reply-To: <o2ya44688a1005051658q204f4507w2e9b118e4289e9ab@mail.gmail.com>
References: <i2ha44688a1005050627n8739d2bcl280c67dddca79866@mail.gmail.com>
	<s2m191c3a031005050940h9e74a96cs237802d4f057b0f2@mail.gmail.com>
	<o2ya44688a1005051658q204f4507w2e9b118e4289e9ab@mail.gmail.com>
Message-ID: <t2n191c3a031005051706za41ed9a6vfdb11c45ac4f9d57@mail.gmail.com>

2g ram in the openwrt box?  isn't it a little plastic toy?

search the wiki for debugging, you would need BT etc and very latest code.


On Wed, May 5, 2010 at 6:58 PM, Woody Dickson <woodydickson at gmail.com>wrote:

> Hi,
>
> I have 2 G of RAM in that box and with 25 calls, there are still
> plenty of memory left.  Is there anyway I can debug this issue?
>
> Are you referring to the astlinux freeswitch package?  I can't seem to
> find it from the astlinux site.
>
> I found a link from some old threads pointing to the astlinux download
> page, but the iso image is not there.
>
> Can someone please point me to the correct download page?
>
> thanks,
> woody
>
>
> On Thu, May 6, 2010 at 12:40 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > maybe you are running out of ram.
> > how many calls do you want on that little toy box?
> > it's already a package there why are you even compiling it.
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
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
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From woodydickson at gmail.com  Wed May  5 17:16:36 2010
From: woodydickson at gmail.com (Woody Dickson)
Date: Thu, 6 May 2010 08:16:36 +0800
Subject: [Freeswitch-users] freeSWITCH segfault on openwrt
In-Reply-To: <t2n191c3a031005051706za41ed9a6vfdb11c45ac4f9d57@mail.gmail.com>
References: <i2ha44688a1005050627n8739d2bcl280c67dddca79866@mail.gmail.com>
	<s2m191c3a031005050940h9e74a96cs237802d4f057b0f2@mail.gmail.com>
	<o2ya44688a1005051658q204f4507w2e9b118e4289e9ab@mail.gmail.com>
	<t2n191c3a031005051706za41ed9a6vfdb11c45ac4f9d57@mail.gmail.com>
Message-ID: <l2xa44688a1005051716wda417209sc62554678b4c8e6e@mail.gmail.com>

Hi

I use gdb, bt, bt full, to open the core file, but it returns an empty
set.  Nothing that points to any memory violation.

I am using 1.0.6 for the ipk that I am building.

What else can I try to fix this problem?

thanks,
woody

On Thu, May 6, 2010 at 8:06 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> 2g ram in the openwrt box? ?isn't it a little plastic toy?
> search the wiki for debugging, you would need BT etc and very latest code.
>
> On Wed, May 5, 2010 at 6:58 PM, Woody Dickson <woodydickson at gmail.com>
> wrote:
>>
>> Hi,
>>
>> I have 2 G of RAM in that box and with 25 calls, there are still
>> plenty of memory left. ?Is there anyway I can debug this issue?
>>
>> Are you referring to the astlinux freeswitch package? ?I can't seem to
>> find it from the astlinux site.
>>
>> I found a link from some old threads pointing to the astlinux download
>> page, but the iso image is not there.
>>
>> Can someone please point me to the correct download page?
>>
>> thanks,
>> woody
>>
>>
>> On Thu, May 6, 2010 at 12:40 AM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>> > maybe you are running out of ram.
>> > how many calls do you want on that little toy box?
>> > it's already a package there why are you even compiling it.
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From asobihoudai at yahoo.com  Wed May  5 17:24:49 2010
From: asobihoudai at yahoo.com (Paul)
Date: Wed, 5 May 2010 17:24:49 -0700 (PDT)
Subject: [Freeswitch-users] freeSWITCH segfault on openwrt
In-Reply-To: <t2n191c3a031005051706za41ed9a6vfdb11c45ac4f9d57@mail.gmail.com>
References: <i2ha44688a1005050627n8739d2bcl280c67dddca79866@mail.gmail.com>
	<s2m191c3a031005050940h9e74a96cs237802d4f057b0f2@mail.gmail.com>
	<o2ya44688a1005051658q204f4507w2e9b118e4289e9ab@mail.gmail.com>
	<t2n191c3a031005051706za41ed9a6vfdb11c45ac4f9d57@mail.gmail.com>
Message-ID: <993032.62970.qm@web111310.mail.gq1.yahoo.com>

I'm assuming that this has already been referenced?

http://wiki.freeswitch.org/wiki/OpenWrt

OpenWRT runs on a wide variety of platforms. While the software is derived from the linux firmware from the Linksys WRT54G boxes, it's been made a lot more versatile by a grip of software developers customizing the firmware to a significant degree. It's now possible to install a lot of enterprise features on one of these embedded devices.



________________________________
From:Anthony Minessale <anthony.minessale at gmail.com>
To:freeswitch-users at lists.freeswitch.org
Sent: Wed, May 5, 2010 5:06:38 PM
Subject: Re: [Freeswitch-users] freeSWITCH segfault on openwrt

2g ram in the openwrt box?  isn't it a little plastic toy?

search the wiki for debugging, you would need BT etc and very latest code.



On Wed, May 5, 2010 at 6:58 PM, Woody Dickson <woodydickson at gmail.com> wrote:

Hi,
>
>>I have 2 G of RAM in that box and with 25 calls, there are still
>>plenty of memory left.  Is there anyway I can debug this issue?
>
>>Are you referring to the astlinux freeswitch package?  I can't seem to
>>find it from the astlinux site.
>
>>I found a link from some old threads pointing to the astlinux download
>>page, but the iso image is not there.
>
>>Can someone please point me to the correct download page?
>
>>thanks,
>woody
>
>
>
>>On Thu, May 6, 2010 at 12:40 AM, Anthony Minessale
>><anthony.minessale at gmail.com> wrote:
>>> maybe you are running out of ram.
>>> how many calls do you want on that little toy box?
>>> it's already a package there why are you even compiling it.
>>>
>
>
>_______________________________________________
>>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>


-- 
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 infos at madovsky.org  Wed May  5 17:35:49 2010
From: infos at madovsky.org (Madovsky)
Date: Wed, 5 May 2010 20:35:49 -0400
Subject: [Freeswitch-users] debug DTMF payload
References: <6D7C014673CA4133908D9691598E6A68@MOBILEE1705>
	<v2j87f2f3b91005051231kca2fbafam2cd3445976445fda@mail.gmail.com>
Message-ID: <5211E70175DC4AD19E950F2A0DA67ED4@MOBILEE1705>

ok thanks
no it's outbound, I'm trying to send rfc2833 packets..

F

  ----- Original Message ----- 
  From: Michael Collins 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, May 05, 2010 3:31 PM
  Subject: Re: [Freeswitch-users] debug DTMF payload


  tcpdump + Wireshark will let you see DTMFs out of band. If you need to see them inband you'll need some sort of software to analyze a recording of the call. IIRC, Audacity has some stuff in it for that.
  -MC


  On Wed, May 5, 2010 at 9:37 AM, Madovsky <infos at madovsky.org> wrote:

    Hi,

    is there any settings in FS that can debug DTMF packets ?
    Or any soft that analyze if it's good or bad DTMF packet.

    Thanks

    F

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






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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From neilp at cs.stanford.edu  Wed May  5 18:13:09 2010
From: neilp at cs.stanford.edu (Neil Patel)
Date: Wed, 5 May 2010 18:13:09 -0700
Subject: [Freeswitch-users] outbound sip call via lua:
	NORMAL_TEMPORARY_FAILURE
Message-ID: <j2we5503bdb1005051813p1444c31al7ef7a2d2bba5c266@mail.gmail.com>

I was testing a luascript that I am using to initiate calls from the command
line using luarun. Here is the relevant snippet of the script:
-- make the call
session = freeswitch.Session(DIALSTRING_PREFIX .. phone_num)
session:setVariable("caller_id_number", phone_num)
session:setVariable("playback_terminators", "#");
session:setHangupHook("hangup");
session:setInputCallback("my_cb", "arg");


This was working fine earlier today, but now I can't get the call to work:
2010-05-05 18:05:46.051908 [NOTICE] sofia.c:4789 Hangup sofia/internal/
sip:1001 at 127.0.0.1:61770 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]
2010-05-05 18:05:46.053759 [ERR] switch_ivr_originate.c:2430 Cannot create
outgoing channel of type [user] cause: [NORMAL_TEMPORARY_FAILURE]
2010-05-05 18:05:46.053759 [ERR] switch_cpp.cpp:604 session is not
initalized
2010-05-05 18:05:46.053759 [ERR] switch_cpp.cpp:604 session is not
initalized
2010-05-05 18:05:46.053759 [ERR] freeswitch_lua.cpp:194 session is not
initalized
2010-05-05 18:05:46.053759 [ERR] freeswitch_lua.cpp:222 session is not
initalized


I can paste more details, but seeing as this was working a short time ago,
I'm wondering whether it is something silly. Thanks in advance.

-Neil
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From brian at freeswitch.org  Wed May  5 18:17:35 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 5 May 2010 20:17:35 -0500
Subject: [Freeswitch-users] outbound sip call via lua:
	NORMAL_TEMPORARY_FAILURE
In-Reply-To: <j2we5503bdb1005051813p1444c31al7ef7a2d2bba5c266@mail.gmail.com>
References: <j2we5503bdb1005051813p1444c31al7ef7a2d2bba5c266@mail.gmail.com>
Message-ID: <7FB00939-BD86-48C1-A203-42CCAB036B23@freeswitch.org>

How are you running this script?

/b

On May 5, 2010, at 8:13 PM, Neil Patel wrote:

> I was testing a luascript that I am using to initiate calls from the command line using luarun. Here is the relevant snippet of the script:
> -- make the call
> session = freeswitch.Session(DIALSTRING_PREFIX .. phone_num)
> session:setVariable("caller_id_number", phone_num)
> session:setVariable("playback_terminators", "#");
> session:setHangupHook("hangup");
> session:setInputCallback("my_cb", "arg");
> 



From jeff at jefflenk.com  Wed May  5 18:46:52 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 5 May 2010 18:46:52 -0700 (PDT)
Subject: [Freeswitch-users] Suggestions for creating diff's in git...
In-Reply-To: <E6DB5478-62E2-4BDF-92A2-75F3CBA84571@jerris.com>
References: <549CFEF87AEDE841A38E9D15EAB4C04C5576CE4D98@cooper>
	<8E40D6E4-05BE-49D4-8BED-BECA792D6B7A@freeswitch.org>
	<549CFEF87AEDE841A38E9D15EAB4C04C5576CE4D99@cooper>
	<v2q7abab2411005031459o36c05f43x4f8ce823a08d9b54@mail.gmail.com>
	<E6DB5478-62E2-4BDF-92A2-75F3CBA84571@jerris.com>
Message-ID: <1273110412885-5012138.post@n2.nabble.com>


Heres what I do if you use TortoiseGit-

Right click file TortoiseGit->ShowLog
select working directory changes
right click "show differences as unified diff"
File "Save as"


-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Suggestions-for-creating-diff-s-in-git-tp5000123p5012138.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From jmesquita at freeswitch.org  Wed May  5 20:05:51 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Thu, 6 May 2010 00:05:51 -0300
Subject: [Freeswitch-users] freeSWITCH segfault on openwrt
In-Reply-To: <l2xa44688a1005051716wda417209sc62554678b4c8e6e@mail.gmail.com>
References: <i2ha44688a1005050627n8739d2bcl280c67dddca79866@mail.gmail.com>
	<s2m191c3a031005050940h9e74a96cs237802d4f057b0f2@mail.gmail.com>
	<o2ya44688a1005051658q204f4507w2e9b118e4289e9ab@mail.gmail.com>
	<t2n191c3a031005051706za41ed9a6vfdb11c45ac4f9d57@mail.gmail.com>
	<l2xa44688a1005051716wda417209sc62554678b4c8e6e@mail.gmail.com>
Message-ID: <w2mebbc2d551005052005oe99951a5w519f406e2959a847@mail.gmail.com>

Just a tip? When Tony asks you for core dump, don't say what you thought of
it without actually providing what he asked. Personally, I also believe that
it would be the right way to have your problem fixed, otherwise, you won't
be having another pair of eyes (or several) looking at information you might
be missing.

Hope that helps as an advice.

Regards,

JM

On Wed, May 5, 2010 at 9:16 PM, Woody Dickson <woodydickson at gmail.com>wrote:

> Hi
>
> I use gdb, bt, bt full, to open the core file, but it returns an empty
> set.  Nothing that points to any memory violation.
>
> I am using 1.0.6 for the ipk that I am building.
>
> What else can I try to fix this problem?
>
> thanks,
> woody
>
> On Thu, May 6, 2010 at 8:06 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > 2g ram in the openwrt box?  isn't it a little plastic toy?
> > search the wiki for debugging, you would need BT etc and very latest
> code.
> >
> > On Wed, May 5, 2010 at 6:58 PM, Woody Dickson <woodydickson at gmail.com>
> > wrote:
> >>
> >> Hi,
> >>
> >> I have 2 G of RAM in that box and with 25 calls, there are still
> >> plenty of memory left.  Is there anyway I can debug this issue?
> >>
> >> Are you referring to the astlinux freeswitch package?  I can't seem to
> >> find it from the astlinux site.
> >>
> >> I found a link from some old threads pointing to the astlinux download
> >> page, but the iso image is not there.
> >>
> >> Can someone please point me to the correct download page?
> >>
> >> thanks,
> >> woody
> >>
> >>
> >> On Thu, May 6, 2010 at 12:40 AM, Anthony Minessale
> >> <anthony.minessale at gmail.com> wrote:
> >> > maybe you are running out of ram.
> >> > how many calls do you want on that little toy box?
> >> > it's already a package there why are you even compiling it.
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > 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-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From moises.silva at gmail.com  Wed May  5 21:27:01 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Thu, 6 May 2010 00:27:01 -0400
Subject: [Freeswitch-users] run error after building in vs 2008!
In-Reply-To: <4BE198AE.7060701@sangoma.com>
References: <i2v1fdf92721004302253l7dbdf254p1ff2b3884ec5a4d4@mail.gmail.com>
	<x2z367751821005011229g3b6c217kbda766daac32a44d@mail.gmail.com>
	<E141A983-9EC5-4C62-8D17-675DC0A67009@jerris.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5576CE4D97@cooper>
	<k2y87f2f3b91005031727j1661d31vf76dc19d9e3354e5@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5576CE4DA1@cooper>
	<n2n1fdf92721005042219x95407e2cp9a0496b95dee3308@mail.gmail.com>
	<n2o47d63d921005042246m6e506ab8s403be7f167229611@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C55777E1C40@cooper>
	<4BE198AE.7060701@sangoma.com>
Message-ID: <u2xc4d05cbe1005052127v8b6373c2z1804d10cf30f13cf@mail.gmail.com>

The FreeTDM instructions are up. We will keep improving them with more
details and screenshots, but this is a start:
http://wiki.sangoma.com/wanpipe-api-freetdm

<http://wiki.sangoma.com/wanpipe-api-freetdm>There is instructions both for
Linux and Windows.

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com


On Wed, May 5, 2010 at 12:11 PM, David Yat Sin <dyatsin at sangoma.com> wrote:

>  Hi Peter,
> No you do not have to purchase NetborderExpress to support Europe ISDN.
>
> You can either use
> 1. Freeswitch + Freetdm + Libpri
> or
> 2. Freeswitch + Freetdm + sangoma_pri stack.
>
> Both options will support EuroISDN variant.
>
> The sangoma_pri stack on Windows is still in testing phase and not
> production ready. We will have instructions on how to deploy it on the
> Sangoma Wiki in a couple hours. I will post the link to the Sangoma wiki
> page as soon as its ready.
>
>  *David Yat Sin, **BEng**, **Software Engineer**
> *Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
> 9T3 Canada
> t. 1 905 474 1990 x 119 | e. *dyatsin at sangoma.com*
>
>
>
> On 5/5/2010 2:54 AM, Peter Olsson wrote:
>
>  This is the link about FS/FreeTDM:
> http://wiki.sangoma.com/wanpipe-api-freetdm
>
>
>
> /Peter
>
>
>
> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [
> mailto:freeswitch-users-bounces at lists.freeswitch.org<freeswitch-users-bounces at lists.freeswitch.org>]
> *F?r *ovvenkat
> *Skickat:* den 5 maj 2010 07:47
> *Till:* freeswitch-users at lists.freeswitch.org
> *?mne:* Re: [Freeswitch-users] run error after building in vs 2008!
>
>
>
> Hi to all,
>
> We have developed a IVR Application in freeSwitch SDK. Now , we want to
> deploye it on the production server. So, we  I bought Sangoma A101 T1/E1
> card. Now we are facing  a new problem like
>
> *FreeSwitch wont support "Europe ISDN" connection*. their technical
> support guys are saying that, we need to buy a additional software called  "
> *NetBorderExpress*" to overcome this problem. Is this true? why its so?
>
> Regards
> Venkatesan OV.
>
> !DSPAM:4be1079832931102718702!
>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From neilp at cs.stanford.edu  Thu May  6 00:12:12 2010
From: neilp at cs.stanford.edu (Neil Patel)
Date: Thu, 6 May 2010 00:12:12 -0700
Subject: [Freeswitch-users] outbound sip call via lua:
	NORMAL_TEMPORARY_FAILURE
In-Reply-To: <7FB00939-BD86-48C1-A203-42CCAB036B23@freeswitch.org>
References: <j2we5503bdb1005051813p1444c31al7ef7a2d2bba5c266@mail.gmail.com> 
	<7FB00939-BD86-48C1-A203-42CCAB036B23@freeswitch.org>
Message-ID: <k2ye5503bdb1005060012vcecc6408l555ffef7505d34fd@mail.gmail.com>

from the CLI:

> luarun script.lua

On Wed, May 5, 2010 at 6:17 PM, Brian West <brian at freeswitch.org> wrote:

> How are you running this script?
>
> /b
>
> On May 5, 2010, at 8:13 PM, Neil Patel wrote:
>
> > I was testing a luascript that I am using to initiate calls from the
> command line using luarun. Here is the relevant snippet of the script:
> > -- make the call
> > session = freeswitch.Session(DIALSTRING_PREFIX .. phone_num)
> > session:setVariable("caller_id_number", phone_num)
> > session:setVariable("playback_terminators", "#");
> > session:setHangupHook("hangup");
> > session:setInputCallback("my_cb", "arg");
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From babak.freeswitch at gmail.com  Thu May  6 01:45:07 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Thu, 6 May 2010 13:15:07 +0430
Subject: [Freeswitch-users] no ring back mod_managed!
Message-ID: <z2n1fdf92721005060145v1a7812a7t6cc89e4c8447ba0d@mail.gmail.com>

Hi
I want to play a file before bridging a call in mod_managed and I'm using
execute_on_answer to begin the recording so I can not use it again
and I'm using this code:
Session.Execute("Playback",m_readyPrompt);
Session.SetVariable("ringback", Session.GetVariable("us_ring"));
   //I've tested this before and after exec of playback but no change
Session.Execute("bridge", string.Format("sofia/internal/{0}%{1}", op,
Session.GetVariable("domain")));

but no ring back is played
Thanks
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From kawarod at laposte.net  Thu May  6 01:59:35 2010
From: kawarod at laposte.net (Rod.)
Date: Thu, 06 May 2010 12:59:35 +0400
Subject: [Freeswitch-users] How to get Context binding for a user
Message-ID: <4BE284F7.3000603@laposte.net>

Hi list,

I'm wondering how to get the context binding for a specific user or 
every user bind in this context.
Tried "sofia_contact" and "sofia status profile #profile_name" but I do 
not have this information.

regards,
rod


From kawarod at laposte.net  Thu May  6 02:13:14 2010
From: kawarod at laposte.net (Rod.)
Date: Thu, 06 May 2010 13:13:14 +0400
Subject: [Freeswitch-users] Registration ODBC feeded
 by	another	registrar proxy
In-Reply-To: <4BDFD6FD.2050903@laposte.net>
References: <4BDED22F.20205@laposte.net>,
	<C5D5A056-B21C-4343-AA13-95D67B3EADA6@gmail.com>,
	<4BDF02AE.3010003@laposte.net>	<SNT135-w18CD702457BE5E08B7D51D2F20@phx.gbl>
	<4BDFD6FD.2050903@laposte.net>
Message-ID: <4BE2882A.7080102@laposte.net>

Hi,

At the moment, I'm registering 40000 users in FS using ODBC with no more 
than 16 registration/sec.
When I do a select in the sip_registrations table, I see that every 
sip_user have an expires value equal to something like "1273141208" even 
if I'm using an expire value of 3600 in my sip register request.

+-----------------------+----------+--------------+----------------+-----------------------------------+-----------------+---------+------------+------------+-------------+---------------+--------------+----------+------------+--------------+--------------+--------------+----------+--------------+------------------+---------------+
| call_id               | sip_user | sip_host     | presence_hosts | 
contact                           | status          | rpid    | 
expires    | user_agent | server_user | server_host   | profile_name | 
hostname | network_ip | network_port | sip_username | sip_realm    | 
mwi_user | mwi_host     | orig_server_host | orig_hostname |
+-----------------------+----------+--------------+----------------+-----------------------------------+-----------------+---------+------------+------------+-------------+---------------+--------------+----------+------------+--------------+--------------+--------------+----------+--------------+------------------+---------------+
| 83755-2574 at 10.10.55.1 | 3755     | 10.10.55.254 | 10.10.55.254   | 
"user" <sip:3755 at 10.10.55.1:5060> | Registered(UDP) | unknown | 
1273141208 | SIPp/Linux | 3755        | 172.30.30.129 | internal     | 
voip     | 10.10.55.1 | 5060         | 3755         | 10.10.55.254 | 
3755     | 10.10.55.254 | 172.30.30.129    | voip          |
+-----------------------+----------+--------------+----------------+-----------------------------------+-----------------+---------+------------+------------+-------------+---------------+--------------+----------+------------+--------------+--------------+--------------+----------+--------------+------------------+---------------+

When I look in the sofia status profile, it seems that the correct 
expires value is set. Any idea why the expires is not set accordingly to 
the SIP REGISTER.

Moreover, when doing a "sofia status profile internal" with such a huge 
subscribers list, the CPU is at 100% and processing no more SIP messages.
Is there a way to prevent this, by setting for example a lower priority 
to CLI request. I already mounted the SQLlite DB in RAM to avoid IO 
bottlenecks.

regards,
rod


Le 04/05/2010 12:12, Rod. a ?crit :
> Hi,
>
> I already thought about that, as I'm using static IP it could be even 
> easier.
> But how to check network connectivity issue and reroute call to 
> voicemail asap: call progress timeout ??
>
> rod
>
>
>
> Le 03/05/2010 21:42, Jan Berger a ?crit :
>> May a suggest a change filter developed if this really is needed?
>>
>> Re-loading everything just in case something has changes is a huge 
>> waste of bandwidth and CPU - if you install an intelligent change 
>> filter you would be down to a few entries changing.
>>
>> Jan
>>
>> ------------------------------------------------------------------------
>> Date: Mon, 3 May 2010 21:06:54 +0400
>> From: kawarod at laposte.net
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Registration ODBC feeded by another 
>> registrar proxy
>>
>> Hi,
>>
>> thanks for your answer and just some details to describe what I'm 
>> looking for.
>> I have to register 25 000 subscribers, no NAT is involved, each 
>> equipment has its own IP address.
>> These equipments are registering every 60 seconds on our current 
>> platform, but I can change this parameter if needed.
>> Equipments are ADSL CPE (router), that's why I'm using 60sec cause 
>> flapping could happen very often with ADSL if the copper line is 
>> crappy. ADSL could be very unpredictable sometimes.
>> As I don't want to delay too much forwarding to voicemail if a user 
>> is unavailable (network issue), 60 sec was chosen (bandwith is not an 
>> issue). But as I told before, I'm open to your suggestions.
>>
>> To Philip, using a single SIP proxy (opensips/ser...) in front of a 
>> FS cluster could be a single point of failure too.
>> I think that maybe a solution using DNS SRV to distribute the load 
>> across a cluster could do the trick or some kind of LVS (virtual IP 
>> shared across many servers)
>> XML curl is a good idea too.
>> To be honest, clustering is a must to avoid a single point of 
>> failure, but FS performance as a SBC are really great even on 
>> commodity hardware, more than 100 CallPerSecond with no transcoding. 
>> That's why I think that a mix with a SIP registrar and FS (and 
>> redundancy) could easily handle my 25 000 subscribers. I did some lab 
>> (one or 2 years ago) with Kamailio registering 90 000 users every 
>> 60sec (1500 Registration per second) without any issues.
>> In my network, 25 000 users are not pushing more than 10 CPS and 500 
>> simultaneous call. I'm not doing VoIP termination.
>>
>> At the moment, I'm just collecting data/feedback on what could be 
>> done as I have some time to work on this project, and if going 
>> further I will share the configuration as I did before:
>> http://wiki.freeswitch.org/wiki/SBC_Setup (not the best setup, but 
>> hope it helps users to begin with FS)
>>
>> regards,
>> rod.
>>
>>
>>
>>
>> Le 03/05/2010 19:54, David Ponzone a ?crit :
>>
>>     Rod,
>>
>>     Registering every 60 seconds is a bad idea, and this should not
>>     be justified.
>>     You should register every 1800 seconds and send a NAT keepalive
>>     every X seconds.
>>     X should be slightly lower than the NAT UDP timeout of the router
>>     in front of the phones, if the phones are behind NAT.
>>     If the phones are not behind NAT, NAT keepalive is not necessary.
>>
>>     David Ponzone Direction Technique
>>     email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>>     tel:      01 74 03 18 97
>>     gsm:   06 66 98 76 34
>>
>>     Service Client IPeva
>>     tel:      0811 46 26 26
>>     www.ipeva.fr  - www.ipeva-studio.com
>>
>>     /Ce message et toutes les pi?ces jointes sont confidentiels et
>>     ?tablis ? l'intention exclusive de ses destinataires. Toute
>>     utilisation ou diffusion non autoris?e est interdite. Tout
>>     message ?lectronique est susceptible d'alt?ration.
>>     /*/IPeva/*/ d?cline toute responsabilit? au titre de ce message
>>     s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>     destinataire de ce message, merci de le d?truire imm?diatement et
>>     d'avertir l'exp?diteur./
>>     /
>>     /
>>
>>
>>
>>     Le 03/05/2010 ? 15:39, Rod. a ?crit :
>>
>>         Hi list,
>>
>>         was playing with FS 1.0.6 and trying to test the registration
>>         performance of FS. (Yes I know FS is more suited as a B2BUA,
>>         but please
>>         read further :p)
>>
>>         So I did the following:
>>             - generate one xml file with 20 000 user account like this:
>>         <include>
>>         <user id="1">
>>         <params>
>>         <param name="password" value="1234"/>
>>         </params>
>>         </user>
>>         <user id="2">
>>         <params>
>>         <param name="password" value="1234"/>
>>         </params>
>>                     ...
>>
>>         Then I used Sipp to test how many registration per second
>>         could be fired
>>         to the server (quad core 2.83Ghz).
>>         I setup ulimit variables, and disable nat.
>>
>>         I got this:
>>             - using SQL Lite: unable to get higher than 80
>>         registrations per
>>         second (in fact it's less than this number but didn't test
>>         too much this
>>         setup), I see a lot of retransmission in Sipp
>>             - using SQL Lite in ramdisk (tmpfs): OK with 80
>>         registrations per
>>         second but not much
>>             - using ODBC and mysql: 130 reg/sec is OK
>>
>>         With ODBC, above 150 reg/sec I see that FS is stalled to
>>         100-110% CPU, I
>>         think it's because I'm using only one SIP profile and that
>>         SOFIA is
>>         monothreaded for this SIP profile.
>>         If I'd like to register every 60sec, the server has to
>>         support at least
>>         more than 300 registration per second.
>>
>>         So I'm wondering if I could setup something like this:
>>             - use another SIP Proxy as a registrar and feed the ODBC
>>         "sip_registration database" of FS
>>             - FS will be able to use this database to setup a call
>>             - use FS as the outbound proxy for call routing
>>
>>         But what about the user params that have been setup in the
>>         xml file
>>         above. I think that FS loads the user params each time a user
>>         is registered.
>>
>>         Comments and advices are welcome.
>>
>>         regards,
>>         rod.
>>
>>
>>
>>         _______________________________________________
>>         FreeSWITCH-users mailing list
>>         FreeSWITCH-users at lists.freeswitch.org
>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>         http://www.freeswitch.org <http://www.freeswitch.org/>
>>
>>
>>
>>     _______________________________________________
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org  <http://www.freeswitch.org/>
>>        
>>
>>
>>
>> ------------------------------------------------------------------------
>> Hotmail: Free, trusted and rich email service. Get it now. 
>> <https://signup.live.com/signup.aspx?id=60969>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>    
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From peter.olsson at visionutveckling.se  Thu May  6 02:14:28 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 6 May 2010 11:14:28 +0200
Subject: [Freeswitch-users] Suggestions for creating diff's in git...
In-Reply-To: <1273110412885-5012138.post@n2.nabble.com>
References: <549CFEF87AEDE841A38E9D15EAB4C04C5576CE4D98@cooper>
	<8E40D6E4-05BE-49D4-8BED-BECA792D6B7A@freeswitch.org>
	<549CFEF87AEDE841A38E9D15EAB4C04C5576CE4D99@cooper>
	<v2q7abab2411005031459o36c05f43x4f8ce823a08d9b54@mail.gmail.com>
	<E6DB5478-62E2-4BDF-92A2-75F3CBA84571@jerris.com>
	<1273110412885-5012138.post@n2.nabble.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C557787689E@cooper>

Jeff, thanks - that's even easier :)

/Peter

-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Jeff Lenk
Skickat: den 6 maj 2010 03:47
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Suggestions for creating diff's in git...


Heres what I do if you use TortoiseGit-

Right click file TortoiseGit->ShowLog
select working directory changes
right click "show differences as unified diff"
File "Save as"


-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Suggestions-for-creating-diff-s-in-git-tp5000123p5012138.html
Sent from the freeswitch-users mailing list archive at Nabble.com.

_______________________________________________
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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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!DSPAM:4be220fe32937827565151!



From pjintheusa at gmail.com  Thu May  6 06:00:25 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Thu, 6 May 2010 09:00:25 -0400
Subject: [Freeswitch-users] Registration ODBC feeded by another
	registrar proxy
In-Reply-To: <4BE2882A.7080102@laposte.net>
References: <4BDED22F.20205@laposte.net>
	<C5D5A056-B21C-4343-AA13-95D67B3EADA6@gmail.com>
	<4BDF02AE.3010003@laposte.net>
	<SNT135-w18CD702457BE5E08B7D51D2F20@phx.gbl>
	<4BDFD6FD.2050903@laposte.net> <4BE2882A.7080102@laposte.net>
Message-ID: <w2y367751821005060600k2a49680cybcb92e12fa3c26aa@mail.gmail.com>

Ron,

I don't know - but I suspect that expires is a actual time. That would make
sense to me - something like:

expires = epochNow + timeout;

As for the 40,000 users - are you really planning doing this on one box?
Registrations + there traffic? Surely doing a FS cluster as discussed or
having OpenSIPS handle registration would be the way to go here.



On Thu, May 6, 2010 at 5:13 AM, Rod. <kawarod at laposte.net> wrote:

>  Hi,
>
> At the moment, I'm registering 40000 users in FS using ODBC with no more
> than 16 registration/sec.
> When I do a select in the sip_registrations table, I see that every
> sip_user have an expires value equal to something like "1273141208" even if
> I'm using an expire value of 3600 in my sip register request.
>
>
> +-----------------------+----------+--------------+----------------+-----------------------------------+-----------------+---------+------------+------------+-------------+---------------+--------------+----------+------------+--------------+--------------+--------------+----------+--------------+------------------+---------------+
> | call_id               | sip_user | sip_host     | presence_hosts |
> contact                           | status          | rpid    | expires    |
> user_agent | server_user | server_host   | profile_name | hostname |
> network_ip | network_port | sip_username | sip_realm    | mwi_user |
> mwi_host     | orig_server_host | orig_hostname |
>
> +-----------------------+----------+--------------+----------------+-----------------------------------+-----------------+---------+------------+------------+-------------+---------------+--------------+----------+------------+--------------+--------------+--------------+----------+--------------+------------------+---------------+
> | 83755-2574 at 10.10.55.1 | 3755     | 10.10.55.254 | 10.10.55.254   |
> "user" <sip:3755 at 10.10.55.1:5060> <sip:3755 at 10.10.55.1:5060> |
> Registered(UDP) | unknown | 1273141208 | SIPp/Linux | 3755        |
> 172.30.30.129 | internal     | voip     | 10.10.55.1 | 5060         |
> 3755         | 10.10.55.254 | 3755     | 10.10.55.254 | 172.30.30.129    |
> voip          |
>
> +-----------------------+----------+--------------+----------------+-----------------------------------+-----------------+---------+------------+------------+-------------+---------------+--------------+----------+------------+--------------+--------------+--------------+----------+--------------+------------------+---------------+
>
> When I look in the sofia status profile, it seems that the correct expires
> value is set. Any idea why the expires is not set accordingly to the SIP
> REGISTER.
>
> Moreover, when doing a "sofia status profile internal" with such a huge
> subscribers list, the CPU is at 100% and processing no more SIP messages.
> Is there a way to prevent this, by setting for example a lower priority to
> CLI request. I already mounted the SQLlite DB in RAM to avoid IO
> bottlenecks.
>
> regards,
> rod
>
>
> Le 04/05/2010 12:12, Rod. a ?crit :
>
> Hi,
>
> I already thought about that, as I'm using static IP it could be even
> easier.
> But how to check network connectivity issue and reroute call to voicemail
> asap: call progress timeout ??
>
> rod
>
>
>
> Le 03/05/2010 21:42, Jan Berger a ?crit :
>
> May a suggest a change filter developed if this really is needed?
>
> Re-loading everything just in case something has changes is a huge waste of
> bandwidth and CPU - if you install an intelligent change filter you would be
> down to a few entries changing.
>
> Jan
>
> ------------------------------
> Date: Mon, 3 May 2010 21:06:54 +0400
> From: kawarod at laposte.net
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Registration ODBC feeded by another
> registrar proxy
>
> Hi,
>
> thanks for your answer and just some details to describe what I'm looking
> for.
> I have to register 25 000 subscribers, no NAT is involved, each equipment
> has its own IP address.
> These equipments are registering every 60 seconds on our current platform,
> but I can change this parameter if needed.
> Equipments are ADSL CPE (router), that's why I'm using 60sec cause flapping
> could happen very often with ADSL if the copper line is crappy. ADSL could
> be very unpredictable sometimes.
> As I don't want to delay too much forwarding to voicemail if a user is
> unavailable (network issue), 60 sec was chosen (bandwith is not an issue).
> But as I told before, I'm open to your suggestions.
>
> To Philip, using a single SIP proxy (opensips/ser...) in front of a FS
> cluster could be a single point of failure too.
> I think that maybe a solution using DNS SRV to distribute the load across a
> cluster could do the trick or some kind of LVS (virtual IP shared across
> many servers)
> XML curl is a good idea too.
> To be honest, clustering is a must to avoid a single point of failure, but
> FS performance as a SBC are really great even on commodity hardware, more
> than 100 CallPerSecond with no transcoding. That's why I think that a mix
> with a SIP registrar and FS (and redundancy) could easily handle my 25 000
> subscribers. I did some lab (one or 2 years ago) with Kamailio registering
> 90 000 users every 60sec (1500 Registration per second) without any issues.
> In my network, 25 000 users are not pushing more than 10 CPS and 500
> simultaneous call. I'm not doing VoIP termination.
>
> At the moment, I'm just collecting data/feedback on what could be done as I
> have some time to work on this project, and if going further I will share
> the configuration as I did before:
> http://wiki.freeswitch.org/wiki/SBC_Setup (not the best setup, but hope it
> helps users to begin with FS)
>
> regards,
> rod.
>
>
>
>
> Le 03/05/2010 19:54, David Ponzone a ?crit :
>
> Rod,
>
>  Registering every 60 seconds is a bad idea, and this should not be
> justified.
> You should register every 1800 seconds and send a NAT keepalive every X
> seconds.
> X should be slightly lower than the NAT UDP timeout of the router in front
> of the phones, if the phones are behind NAT.
> If the phones are not behind NAT, NAT keepalive is not necessary.
>
>   David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
>  Service Client IPeva
>  tel:      0811 46 26 26
>  www.ipeva.fr  -   www.ipeva-studio.com
>
>  *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
>  Le 03/05/2010 ? 15:39, Rod. a ?crit :
>
>  Hi list,
>
> was playing with FS 1.0.6 and trying to test the registration
> performance of FS. (Yes I know FS is more suited as a B2BUA, but please
> read further :p)
>
> So I did the following:
>     - generate one xml file with 20 000 user account like this:
> <include>
> <user id="1">
> <params>
> <param name="password" value="1234"/>
> </params>
> </user>
> <user id="2">
> <params>
> <param name="password" value="1234"/>
> </params>
>             ...
>
> Then I used Sipp to test how many registration per second could be fired
> to the server (quad core 2.83Ghz).
> I setup ulimit variables, and disable nat.
>
> I got this:
>     - using SQL Lite: unable to get higher than 80 registrations per
> second (in fact it's less than this number but didn't test too much this
> setup), I see a lot of retransmission in Sipp
>     - using SQL Lite in ramdisk (tmpfs): OK with 80 registrations per
> second but not much
>     - using ODBC and mysql: 130 reg/sec is OK
>
> With ODBC, above 150 reg/sec I see that FS is stalled to 100-110% CPU, I
> think it's because I'm using only one SIP profile and that SOFIA is
> monothreaded for this SIP profile.
> If I'd like to register every 60sec, the server has to support at least
> more than 300 registration per second.
>
> So I'm wondering if I could setup something like this:
>     - use another SIP Proxy as a registrar and feed the ODBC
> "sip_registration database" of FS
>     - FS will be able to use this database to setup a call
>     - use FS as the outbound proxy for call routing
>
> But what about the user params that have been setup in the xml file
> above. I think that FS loads the user params each time a user is
> registered.
>
> Comments and advices are welcome.
>
> regards,
> rod.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> ------------------------------
> Hotmail: Free, trusted and rich email service. Get it now.<https://signup.live.com/signup.aspx?id=60969>
>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From pjintheusa at gmail.com  Thu May  6 06:29:58 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Thu, 6 May 2010 09:29:58 -0400
Subject: [Freeswitch-users] Getting the IP (domain) of the FS box a
	phone registered with
In-Reply-To: <o2g367751821005040724p5fc0d0acyed885ac8cf9816bb@mail.gmail.com>
References: <g2i367751821005031753u16511e8ez684f47c7e7f8cf0f@mail.gmail.com>
	<4BDFBCAF.2020108@epbx.cz>
	<66CEF6A4-2E4D-477E-8BE5-37033BC3F1FF@freeswitch.org>
	<w2s367751821005040538sdfabcbf4wc9b1f5340d98dea7@mail.gmail.com>
	<i2z59ad9ca11005040604kf8ea18dckd99a842fbfc0966c@mail.gmail.com>
	<o2v367751821005040615z3537eec3s8896fa8db263074e@mail.gmail.com>
	<m2p59ad9ca11005040650rbb5bd03gd854fc15047c5330@mail.gmail.com>
	<o2g367751821005040724p5fc0d0acyed885ac8cf9816bb@mail.gmail.com>
Message-ID: <j2g367751821005060629td44ba3d8l73522426f5f02d9d@mail.gmail.com>

The answer here is that FS_PATH does support sending a call through another
FS box.

For example - if a user is registered on FS2 and the call arrives on FS1 -
you can use FS_PATH to route the call through FS2.

sophia/internal/7625558722 at 64.76.25.998;fs_path=sip:64.76.25.999

No changes to the dialplan / scripting is required on FS2.

Note that randomly trying to route with fs_path through any FS box (use it a
pure proxy) will not work.

Also using user/7625558722 at 64.76.25.998;fs_path=sip:64.76.25.999 will also
not work correctly.


On Tue, May 4, 2010 at 10:24 AM, Phillip Jones <pjintheusa at gmail.com> wrote:

> Henry - I don't know - but I will be trying this this afternoon so will let
> you know.
>
> Now if I could only find the FS box that a phone is registered with - I
> would be golden!
>
>
>
> On Tue, May 4, 2010 at 9:50 AM, Henry Huang <red.rain.seven at gmail.com>wrote:
>
>> Phillip:
>>
>> Thanks. Does the proxy need to be configured in some way to be used as a
>> path? For instance, if the proxy being used as path is another FS. Does the
>> FS need to do anything special in the dialplan ? or it's a sofia level
>> thing?
>>
>> Henry
>>
>>
>> On Tue, May 4, 2010 at 9:15 PM, Phillip Jones <pjintheusa at gmail.com>wrote:
>>
>>>
>>> http://wiki.freeswitch.org/wiki/Mod_sofia#Specifying_SIP_Proxy_With_fs_path
>>>
>>>
>>> On Tue, May 4, 2010 at 9:04 AM, Henry Huang <red.rain.seven at gmail.com>wrote:
>>>
>>>> Brian:
>>>>
>>>> Where can I find information regarding FreeSWITCH path?
>>>>
>>>> Thanks,
>>>>
>>>> Henry
>>>>
>>>>
>>>> On Tue, May 4, 2010 at 8:38 PM, Phillip Jones <pjintheusa at gmail.com>wrote:
>>>>
>>>>> kokoska,
>>>>>
>>>>> thanks for that. I am considering setting up opensips to deal with
>>>>> mobile phone connections (as well as carrier connections) but it just
>>>>> introduces a single point of failure. I am trying to ing to avoid this.
>>>>>
>>>>>
>>>>> At this point I am wondering whether there is a simple way of finding
>>>>> out which FS box a client is registered with. (without doing a SQL query
>>>>> into the shared DB - although that would be ok too)
>>>>>
>>>>> I am wondering then whether I can use FS_PATH to route through that box
>>>>> (domain)
>>>>>
>>>>> Thanks!
>>>>>
>>>>> Pj
>>>>>
>>>>>
>>>>>
>>>>> On Tue, May 4, 2010 at 2:51 AM, Brian West <brian at freeswitch.org>wrote:
>>>>>
>>>>>> We already have path support in FreeSWITCH.
>>>>>>
>>>>>> /b
>>>>>>
>>>>>>
>>>>>> On May 4, 2010, at 1:20 AM, Martin Dvorak <martin at epbx.cz> wrote:
>>>>>>
>>>>>> > Dne 4.5.2010 2:53, Phillip Jones napsal(a):
>>>>>> >> Hi there,
>>>>>> >>
>>>>>> >> I have a cluster of FS boxes.  OpenSIPS sits in front of the boxes
>>>>>> to
>>>>>> >> load balance.
>>>>>> >>
>>>>>> >> The internal profile is sharing the same DB via ODBC.
>>>>>> >>
>>>>>> >> A mobile phone SIP client is registering with FS1, and a call for
>>>>>> that
>>>>>> >> SIP client arrives on FS2. The mobile phone will not accept
>>>>>> unsolicited
>>>>>> >> IP traffic, so FS1 must send the invite.
>>>>>> >>
>>>>>> >> From FS2, how do I find the IP (domain) the mobile phone is
>>>>>> registered
>>>>>> >> to? Without iterating through all my domains with sofia_contact? Or
>>>>>> >> doing a look up in the DB.
>>>>>> >>
>>>>>> >>
>>>>>> >> Is there a simpler way?
>>>>>> >>
>>>>>> >
>>>>>> > I think yes :-)
>>>>>> >
>>>>>> > If your phones registering "through" OpenSIPS (OpenSIPS just
>>>>>> forwards
>>>>>> > REGISTERs to FreeSWITCH boxes), they should accept traffic from
>>>>>> OpenSIPS
>>>>>> > IP address.
>>>>>> > And than you could send calls back to your phones through OpenSIPS
>>>>>> too.
>>>>>> > "Path" support is needed on FreeSWITCH side, but I hope it works
>>>>>> (but I
>>>>>> > tested it year ago or more).
>>>>>> >
>>>>>> > Beste regards,
>>>>>> >
>>>>>> > kokoska.rokoska
>>>>>> >
>>>>>> > _______________________________________________
>>>>>> > FreeSWITCH-users mailing list
>>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> > UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> > http://www.freeswitch.org
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Henry Huang
>>>> UniC Solution - Communication Unified
>>>> VoIP & Open Source software Consultant
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Henry Huang
>> UniC Solution - Communication Unified
>> VoIP & Open Source software Consultant
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From david.varnes at gmail.com  Thu May  6 07:28:10 2010
From: david.varnes at gmail.com (david varnes)
Date: Fri, 7 May 2010 00:28:10 +1000
Subject: [Freeswitch-users] Can I configure sofia to send an ESL event when
	REGISTER auth *fails* ?
Message-ID: <m2n74a861001005060728xc948fbdcg9a23576a3a75d80@mail.gmail.com>

Hi list,

I have listened for "all" events, and the custom "sofia:register" events,
but I do not seem to get any event when an auth REGISTER fails.

I am trying to get all the info (username, ip address, user-agent etc)
that is in the REGISTER event, but for the failed case as well.

Thanks for any pointers,

davidv

-- 
david varnes

e: david.varnes at gmail.com
p: +61 404 925 633


From brian at freeswitch.org  Thu May  6 07:34:18 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 6 May 2010 09:34:18 -0500
Subject: [Freeswitch-users] Can I configure sofia to send an ESL event
	when REGISTER auth *fails* ?
In-Reply-To: <m2n74a861001005060728xc948fbdcg9a23576a3a75d80@mail.gmail.com>
References: <m2n74a861001005060728xc948fbdcg9a23576a3a75d80@mail.gmail.com>
Message-ID: <9E5C87E6-4D52-426E-BF32-0E61C7C217C2@freeswitch.org>

Are you trying to do fail2ban?

/b

On May 6, 2010, at 9:28 AM, david varnes wrote:

> I am trying to get all the info (username, ip address, user-agent etc)
> that is in the REGISTER event, but for the failed case as well.
> 
> Thanks for any pointers,
> 
> davidv



From max.clark at gmail.com  Thu May  6 07:36:04 2010
From: max.clark at gmail.com (Max Clark)
Date: Thu, 6 May 2010 07:36:04 -0700
Subject: [Freeswitch-users] 4x T1 PRI Gateway
Message-ID: <p2s2fa1e1781005060736u41f4d0d9w1efa59f8a28172fa@mail.gmail.com>

Hello,

I am looking for a 4 port T1/PRI to SIP gateway to interconnect a
legacy PBX to a SIP trunk. After spending some time with Google I have
come up with the following list of devices that fit the required
number of T1 interfaces:

Dialogic DMG2120DTI
Cisco AS5350
Quintum Tenor DX
Patton SmartNode 4960
Audiocodes Mediant 1000/2000

Any opinions, personal experiences, advice, or additional devices I
should be looking at would be greatly appreciated.

Best,
Max


From imthiyaz at peopletech.co.in  Thu May  6 07:46:25 2010
From: imthiyaz at peopletech.co.in (Imthiyaz Ahmed)
Date: Thu, 6 May 2010 20:16:25 +0530
Subject: [Freeswitch-users] 4x T1 PRI Gateway
In-Reply-To: <p2s2fa1e1781005060736u41f4d0d9w1efa59f8a28172fa@mail.gmail.com>
References: <p2s2fa1e1781005060736u41f4d0d9w1efa59f8a28172fa@mail.gmail.com>
Message-ID: <n2n8595daf71005060746zd2b64554uceadb5917ae8d1c1@mail.gmail.com>

Try sangoma Netborder Express , a very cost effective solution and work well
in linux too

On Thu, May 6, 2010 at 8:06 PM, Max Clark <max.clark at gmail.com> wrote:

> Hello,
>
> I am looking for a 4 port T1/PRI to SIP gateway to interconnect a
> legacy PBX to a SIP trunk. After spending some time with Google I have
> come up with the following list of devices that fit the required
> number of T1 interfaces:
>
> Dialogic DMG2120DTI
> Cisco AS5350
> Quintum Tenor DX
> Patton SmartNode 4960
> Audiocodes Mediant 1000/2000
>
> Any opinions, personal experiences, advice, or additional devices I
> should be looking at would be greatly appreciated.
>
> Best,
> Max
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Best Regards
G.Imthiyaz Ahmed
PeopleTech systems (P) ltd
http://peopletech.co.in
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From david.varnes at gmail.com  Thu May  6 07:55:06 2010
From: david.varnes at gmail.com (david varnes)
Date: Fri, 7 May 2010 00:55:06 +1000
Subject: [Freeswitch-users] Can I configure sofia to send an ESL event
	when REGISTER auth *fails* ?
In-Reply-To: <9E5C87E6-4D52-426E-BF32-0E61C7C217C2@freeswitch.org>
References: <m2n74a861001005060728xc948fbdcg9a23576a3a75d80@mail.gmail.com>
	<9E5C87E6-4D52-426E-BF32-0E61C7C217C2@freeswitch.org>
Message-ID: <t2q74a861001005060755p502f87b6n3333196f1735ba33@mail.gmail.com>

Brian,
Thanks for your reply.

No. This is a project on a closed, trusted network and I want to
do some custom phone detection / provisioning for devices that
may have wrong config.

There is one case where the only way I know they exist is because
they try to register (with incorrect credentials).  The combination
of ip address and user-agent would be perfect to allow the correct
config to get pushed to them.

regards
davidv


On 7 May 2010 00:34, Brian West <brian at freeswitch.org> wrote:
> Are you trying to do fail2ban?
>
> /b
>
> On May 6, 2010, at 9:28 AM, david varnes wrote:
>
>> I am trying to get all the info (username, ip address, user-agent etc)
>> that is in the REGISTER event, but for the failed case as well.
>>
>> Thanks for any pointers,
>>
>> davidv
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
david varnes

e: david.varnes at gmail.com
p: +61 404 925 633


From brian at freeswitch.org  Thu May  6 07:55:41 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 6 May 2010 09:55:41 -0500
Subject: [Freeswitch-users] 4x T1 PRI Gateway
In-Reply-To: <n2n8595daf71005060746zd2b64554uceadb5917ae8d1c1@mail.gmail.com>
References: <p2s2fa1e1781005060736u41f4d0d9w1efa59f8a28172fa@mail.gmail.com>
	<n2n8595daf71005060746zd2b64554uceadb5917ae8d1c1@mail.gmail.com>
Message-ID: <1BB557BC-E029-4A2C-89CA-2C94B73739F9@freeswitch.org>

Or FreeSWITCH+FreeTDM on Sangoma... works well too.

/b

On May 6, 2010, at 9:46 AM, Imthiyaz Ahmed wrote:

> Try sangoma Netborder Express , a very cost effective solution and work well in linux too



From tjardick at vanderkraan.net  Thu May  6 07:59:52 2010
From: tjardick at vanderkraan.net (Tjardick van der Kraan)
Date: Thu, 6 May 2010 16:59:52 +0200
Subject: [Freeswitch-users] 4x T1 PRI Gateway
In-Reply-To: <p2s2fa1e1781005060736u41f4d0d9w1efa59f8a28172fa@mail.gmail.com>
References: <p2s2fa1e1781005060736u41f4d0d9w1efa59f8a28172fa@mail.gmail.com>
Message-ID: <8A5AC4E3-EA15-4CCF-B324-873CFD342B78@vanderkraan.net>

Hi Max,

We've been having good results with the Aculab ApplianX gateways.

Regards,

Tjardick


On 06 May 2010, at 16:36, Max Clark wrote:

> Hello,
> 
> I am looking for a 4 port T1/PRI to SIP gateway to interconnect a
> legacy PBX to a SIP trunk. After spending some time with Google I have
> come up with the following list of devices that fit the required
> number of T1 interfaces:
> 
> Dialogic DMG2120DTI
> Cisco AS5350
> Quintum Tenor DX
> Patton SmartNode 4960
> Audiocodes Mediant 1000/2000
> 
> Any opinions, personal experiences, advice, or additional devices I
> should be looking at would be greatly appreciated.
> 
> Best,
> Max
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From infos at madovsky.org  Thu May  6 08:36:17 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 6 May 2010 11:36:17 -0400
Subject: [Freeswitch-users] rfc2833 confusion
Message-ID: <85F48B9892404407AD528655D9414915@MOBILEE1705>

HI,

is rfc2833 inband or outband ? is it only event or mixed event and sound ?
I'm confused after read articles on this subject

Thx

F
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From anthony.minessale at gmail.com  Thu May  6 08:40:51 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 6 May 2010 10:40:51 -0500
Subject: [Freeswitch-users] FreeSWITCH 1.2 (Dragon) on the Horizon Branch
	Maintainers Wanted
Message-ID: <u2g191c3a031005060840p8efb5ed7o10947d4d4e2703a6@mail.gmail.com>

This month marks 2 years since the release of FreeSWITCH 1.0 (Phoenix)
We have added countless features during that time and have held back from
updating some or dependancies for stability concerns.
The time has come to advance FreeSWITCH to the 1.2 version (Dragon)
We would like to branch 1.0 to preserve it but we need volunteers to
maintain it.

Please contact consulting at freeswitch.org or bug us on IRC if you
are interested in helping to maintain the 1.0 branch.


-- 
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
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From janvb at live.com  Thu May  6 08:49:10 2010
From: janvb at live.com (Jan Berger)
Date: Thu, 6 May 2010 17:49:10 +0200
Subject: [Freeswitch-users] 4x T1 PRI Gateway
In-Reply-To: <p2s2fa1e1781005060736u41f4d0d9w1efa59f8a28172fa@mail.gmail.com>
References: <p2s2fa1e1781005060736u41f4d0d9w1efa59f8a28172fa@mail.gmail.com>
Message-ID: <SNT135-w1218CD42C8476DCF66433FD2F50@phx.gbl>


What signalling protocol will you use to connect to the legacy PBX and how much work are you prepard to do yourself? It's critical that you get exact information on your legacy PBX signalling

 

- R1

- R2

- DMS

- AT&T

- ANSI ISDN

- SS7

etc...

 

In general - FreeSWITCH can do it with a medium PC and a 4 x T1 card from Sangoma etc. The cost for a 4xT1 solution will be significantly cheaper than anything than a so-called plug & play box.

 

Jan


 
> Date: Thu, 6 May 2010 07:36:04 -0700
> From: max.clark at gmail.com
> To: Freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] 4x T1 PRI Gateway
> 
> Hello,
> 
> I am looking for a 4 port T1/PRI to SIP gateway to interconnect a
> legacy PBX to a SIP trunk. After spending some time with Google I have
> come up with the following list of devices that fit the required
> number of T1 interfaces:
> 
> Dialogic DMG2120DTI
> Cisco AS5350
> Quintum Tenor DX
> Patton SmartNode 4960
> Audiocodes Mediant 1000/2000
> 
> Any opinions, personal experiences, advice, or additional devices I
> should be looking at would be greatly appreciated.
> 
> Best,
> Max
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  
_________________________________________________________________
Hotmail: Powerful Free email with security by Microsoft.
https://signup.live.com/signup.aspx?id=60969
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From anthony.minessale at gmail.com  Thu May  6 08:52:35 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 6 May 2010 10:52:35 -0500
Subject: [Freeswitch-users] rfc2833 confusion
In-Reply-To: <85F48B9892404407AD528655D9414915@MOBILEE1705>
References: <85F48B9892404407AD528655D9414915@MOBILEE1705>
Message-ID: <u2v191c3a031005060852nd07c7bfbgb399440777f5db89@mail.gmail.com>

both 2833 and actual tones in the audio stream are often called in-band
because the 2833 is just a transport for the inband tones that is assumed to
be reconstructed on the other side.  Some people refer to inband as the real
tones and 2833 as not inband because it's not the tones.  So the one you
have to look out for is people who call 2833 inband and do not clarify that
it's inband packetized according the RFC2833

The one that is not inband for sure is INFO (which has been dropped from the
standard in favor of some ridiculous SUBSCRIBE based xml exchange)

Like all things in SIP and anything else that comes with an RFC even if you
know the answer you can't rely on trusting that answer because the real
world will do whatever it wants and invalidate you. =D


On Thu, May 6, 2010 at 10:36 AM, Madovsky <infos at madovsky.org> wrote:

>  HI,
>
> is rfc2833 inband or outband ? is it only event or mixed event and sound ?
> I'm confused after read articles on this subject
>
> Thx
>
> F
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <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
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From tjardick at vanderkraan.net  Thu May  6 08:55:14 2010
From: tjardick at vanderkraan.net (Tjardick van der Kraan)
Date: Thu, 6 May 2010 17:55:14 +0200
Subject: [Freeswitch-users] rfc2833 confusion
In-Reply-To: <85F48B9892404407AD528655D9414915@MOBILEE1705>
References: <85F48B9892404407AD528655D9414915@MOBILEE1705>
Message-ID: <321621B3-416A-429B-9791-6E507AD1EBFA@vanderkraan.net>

Hi,

rfc2833 DTMF is out-of-band e.g. it is not within the audio of the call, but it does transport thru RTP.

Regards,

Tjardick

On 06 May 2010, at 17:36, Madovsky wrote:

> HI,
>  
> is rfc2833 inband or outband ? is it only event or mixed event and sound ?
> I'm confused after read articles on this subject
>  
> Thx
>  
> F
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Thu May  6 09:00:06 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 6 May 2010 12:00:06 -0400
Subject: [Freeswitch-users] alpha-numeric password
In-Reply-To: <4BE1646F.2060601@mdsh.com>
References: <4BE00C5A.6090201@mdsh.com>
	<AANLkTimWn0SSDF8gbwtu4CL6MSYaQ8fQ61p4ixUONm9v@mail.gmail.com>
	<4BE1646F.2060601@mdsh.com>
Message-ID: <E715555A-E307-44CE-A474-14C3CC0CC2CA@jerris.com>

fs_cli -x reloadxml 

On May 5, 2010, at 8:28 AM, Mark Himsley wrote:

> Thanks for your unequivocal answer.
> 
> FS was sent a HUP signal with `sudo /etc/init.d/freeswitch reload` with
> the standard FS apt-get install Ubuntu server 9.10.
> 
> I now realise that this command was not reloading
> ${FREESWITCH_CONF}/directory/default/201.xml and I needed to do a
> restart instead.
> 
> I can see that I need to get to the freeswitch command line (like *'s
> -r) for better configuring/debugging so I'll look into running FS in
> screen, hopefully without destroying the Ubuntu standard init script too
> much.
> 
> Thanks. Your unequivocal answer made me do a lot more debugging :-)
> 
> 
> On 04/05/2010 17:48, Anthony Minessale wrote:
>> There is no such limitation
>> did you do reloadxml and or restart FS after you changed the configuration?



From infos at madovsky.org  Thu May  6 09:35:58 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 6 May 2010 12:35:58 -0400
Subject: [Freeswitch-users] rfc2833 confusion
References: <85F48B9892404407AD528655D9414915@MOBILEE1705>
	<321621B3-416A-429B-9791-6E507AD1EBFA@vanderkraan.net>
Message-ID: <1654930B4DC94E9FB10C8481C69F53B1@MOBILEE1705>

Thanks Tjardick.
so if I understand rfc2833 is only event ?
i'm trying to send it through RTP as pt=101 but no success until now.
does it need any redundant packet ? confused also if I need to send string structure or object that represent the string vars in RTP flow...

Thanks
  ----- Original Message ----- 
  From: Tjardick van der Kraan 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, May 06, 2010 11:55 AM
  Subject: Re: [Freeswitch-users] rfc2833 confusion


  Hi,


  rfc2833 DTMF is out-of-band e.g. it is not within the audio of the call, but it does transport thru RTP.


  Regards,


  Tjardick


  On 06 May 2010, at 17:36, Madovsky wrote:


    HI,

    is rfc2833 inband or outband ? is it only event or mixed event and sound ?
    I'm confused after read articles on this subject

    Thx

    F
    _______________________________________________
    FreeSWITCH-users mailing list
    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From infos at madovsky.org  Thu May  6 09:39:41 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 6 May 2010 12:39:41 -0400
Subject: [Freeswitch-users] rfc2833 confusion
References: <85F48B9892404407AD528655D9414915@MOBILEE1705>
	<u2v191c3a031005060852nd07c7bfbgb399440777f5db89@mail.gmail.com>
Message-ID: <572512C4F4A540638ACD07DE3631D159@MOBILEE1705>

god, so many confusion for only an audio tone ! ;)
anyway if I understand it's called inband by the fact that it's in RTP flow onlly.
so the rfc2833 packet is only an event payload ...

  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, May 06, 2010 11:52 AM
  Subject: Re: [Freeswitch-users] rfc2833 confusion


  both 2833 and actual tones in the audio stream are often called in-band because the 2833 is just a transport for the inband tones that is assumed to be reconstructed on the other side.  Some people refer to inband as the real tones and 2833 as not inband because it's not the tones.  So the one you have to look out for is people who call 2833 inband and do not clarify that it's inband packetized according the RFC2833 


  The one that is not inband for sure is INFO (which has been dropped from the standard in favor of some ridiculous SUBSCRIBE based xml exchange) 


  Like all things in SIP and anything else that comes with an RFC even if you know the answer you can't rely on trusting that answer because the real world will do whatever it wants and invalidate you. =D




  On Thu, May 6, 2010 at 10:36 AM, Madovsky <infos at madovsky.org> wrote:

    HI,

    is rfc2833 inband or outband ? is it only event or mixed event and sound ?
    I'm confused after read articles on this subject

    Thx

    F

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





  -- 
  Anthony Minessale II

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

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

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



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From infos at madovsky.org  Thu May  6 10:25:11 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 6 May 2010 13:25:11 -0400
Subject: [Freeswitch-users] rfc2833 confusion
Message-ID: <EC805A0F52A240E288F27CBD02317FB6@MOBILEE1705>

I anyone knows a link of a programming example of how
to inject rfc2833 in RTP flow (doesn't matter the language)
It would be very useful

Thanks

F
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, May 06, 2010 12:39 PM
  Subject: Re: [Freeswitch-users] rfc2833 confusion


  god, so many confusion for only an audio tone ! ;)
  anyway if I understand it's called inband by the fact that it's in RTP flow onlly.
  so the rfc2833 packet is only an event payload ...

    ----- Original Message ----- 
    From: Anthony Minessale 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Thursday, May 06, 2010 11:52 AM
    Subject: Re: [Freeswitch-users] rfc2833 confusion


    both 2833 and actual tones in the audio stream are often called in-band because the 2833 is just a transport for the inband tones that is assumed to be reconstructed on the other side.  Some people refer to inband as the real tones and 2833 as not inband because it's not the tones.  So the one you have to look out for is people who call 2833 inband and do not clarify that it's inband packetized according the RFC2833  


    The one that is not inband for sure is INFO (which has been dropped from the standard in favor of some ridiculous SUBSCRIBE based xml exchange) 


    Like all things in SIP and anything else that comes with an RFC even if you know the answer you can't rely on trusting that answer because the real world will do whatever it wants and invalidate you. =D




    On Thu, May 6, 2010 at 10:36 AM, Madovsky <infos at madovsky.org> wrote:

      HI,

      is rfc2833 inband or outband ? is it only event or mixed event and sound ?
      I'm confused after read articles on this subject

      Thx

      F

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





    -- 
    Anthony Minessale II

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

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

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



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


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From pjintheusa at gmail.com  Thu May  6 10:43:40 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Thu, 6 May 2010 13:43:40 -0400
Subject: [Freeswitch-users] rfc2833 confusion
In-Reply-To: <EC805A0F52A240E288F27CBD02317FB6@MOBILEE1705>
References: <EC805A0F52A240E288F27CBD02317FB6@MOBILEE1705>
Message-ID: <v2o367751821005061043hccd410bbxc278154fa42d703a@mail.gmail.com>

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

Don't get to hung up on the inband / out of band thing. This will be
negotiated by FS and the carrier when the call is set up. For instance, when
you use G729 you need out of band DTMF but for G711 you can use both. This
might change from call you call, depending on what media gateways can
handle.

If you want to "see" and hear DTMF - fire up wireshark (or Tshark) and
capture the trace. You can see the RTP DTMF in the trace and hear the
clamped audio from the carrier perhaps. Or just hear the inband tones. It
gives a good visual of what is going on.

On Thu, May 6, 2010 at 1:25 PM, Madovsky <infos at madovsky.org> wrote:

>  I anyone knows a link of a programming example of how
> to inject rfc2833 in RTP flow (doesn't matter the language)
> It would be very useful
>
> Thanks
>
> F
>
> ----- Original Message -----
> *From:* Madovsky <infos at madovsky.org>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Thursday, May 06, 2010 12:39 PM
> *Subject:* Re: [Freeswitch-users] rfc2833 confusion
>
> god, so many confusion for only an audio tone ! ;)
> anyway if I understand it's called inband by the fact that it's in RTP flow
> onlly.
> so the rfc2833 packet is only an event payload ...
>
>
> ----- Original Message -----
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Thursday, May 06, 2010 11:52 AM
> *Subject:* Re: [Freeswitch-users] rfc2833 confusion
>
> both 2833 and actual tones in the audio stream are often called in-band
> because the 2833 is just a transport for the inband tones that is assumed to
> be reconstructed on the other side.  Some people refer to inband as the real
> tones and 2833 as not inband because it's not the tones.  So the one you
> have to look out for is people who call 2833 inband and do not clarify that
> it's inband packetized according the RFC2833
>
> The one that is not inband for sure is INFO (which has been dropped from
> the standard in favor of some ridiculous SUBSCRIBE based xml exchange)
>
> Like all things in SIP and anything else that comes with an RFC even if you
> know the answer you can't rely on trusting that answer because the real
> world will do whatever it wants and invalidate you. =D
>
>
> On Thu, May 6, 2010 at 10:36 AM, Madovsky <infos at madovsky.org> wrote:
>
>>  HI,
>>
>> is rfc2833 inband or outband ? is it only event or mixed event and sound ?
>> I'm confused after read articles on this subject
>>
>> Thx
>>
>> F
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Thu May  6 11:06:58 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 6 May 2010 14:06:58 -0400
Subject: [Freeswitch-users] rfc2833 confusion
References: <EC805A0F52A240E288F27CBD02317FB6@MOBILEE1705>
	<v2o367751821005061043hccd410bbxc278154fa42d703a@mail.gmail.com>
Message-ID: <3288629D99F649B7A222E6CB90CAEFE4@MOBILEE1705>

perfect. thanks !
  ----- Original Message ----- 
  From: Phillip Jones 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, May 06, 2010 1:43 PM
  Subject: Re: [Freeswitch-users] rfc2833 confusion


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

  Don't get to hung up on the inband / out of band thing. This will be negotiated by FS and the carrier when the call is set up. For instance, when you use G729 you need out of band DTMF but for G711 you can use both. This might change from call you call, depending on what media gateways can handle.

  If you want to "see" and hear DTMF - fire up wireshark (or Tshark) and capture the trace. You can see the RTP DTMF in the trace and hear the clamped audio from the carrier perhaps. Or just hear the inband tones. It gives a good visual of what is going on.


  On Thu, May 6, 2010 at 1:25 PM, Madovsky <infos at madovsky.org> wrote:

    I anyone knows a link of a programming example of how
    to inject rfc2833 in RTP flow (doesn't matter the language)
    It would be very useful

    Thanks

    F
      ----- Original Message ----- 
      From: Madovsky 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Thursday, May 06, 2010 12:39 PM
      Subject: Re: [Freeswitch-users] rfc2833 confusion


      god, so many confusion for only an audio tone ! ;)
      anyway if I understand it's called inband by the fact that it's in RTP flow onlly.
      so the rfc2833 packet is only an event payload ...

        ----- Original Message ----- 
        From: Anthony Minessale 
        To: freeswitch-users at lists.freeswitch.org 
        Sent: Thursday, May 06, 2010 11:52 AM
        Subject: Re: [Freeswitch-users] rfc2833 confusion


        both 2833 and actual tones in the audio stream are often called in-band because the 2833 is just a transport for the inband tones that is assumed to be reconstructed on the other side.  Some people refer to inband as the real tones and 2833 as not inband because it's not the tones.  So the one you have to look out for is people who call 2833 inband and do not clarify that it's inband packetized according the RFC2833  


        The one that is not inband for sure is INFO (which has been dropped from the standard in favor of some ridiculous SUBSCRIBE based xml exchange) 


        Like all things in SIP and anything else that comes with an RFC even if you know the answer you can't rely on trusting that answer because the real world will do whatever it wants and invalidate you. =D




        On Thu, May 6, 2010 at 10:36 AM, Madovsky <infos at madovsky.org> wrote:

          HI,

          is rfc2833 inband or outband ? is it only event or mixed event and sound ?
          I'm confused after read articles on this subject

          Thx

          F

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





        -- 
        Anthony Minessale II

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

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

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



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


        _______________________________________________
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        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From infos at madovsky.org  Thu May  6 11:10:47 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 6 May 2010 14:10:47 -0400
Subject: [Freeswitch-users] rfc2833 confusion
References: <EC805A0F52A240E288F27CBD02317FB6@MOBILEE1705>
	<v2o367751821005061043hccd410bbxc278154fa42d703a@mail.gmail.com>
Message-ID: <3B1B689F043C4AEC8804FEED82AD153B@MOBILEE1705>

But how to catch digits form a dialplan when the call
is already on ?
  ----- Original Message ----- 
  From: Phillip Jones 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, May 06, 2010 1:43 PM
  Subject: Re: [Freeswitch-users] rfc2833 confusion


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

  Don't get to hung up on the inband / out of band thing. This will be negotiated by FS and the carrier when the call is set up. For instance, when you use G729 you need out of band DTMF but for G711 you can use both. This might change from call you call, depending on what media gateways can handle.

  If you want to "see" and hear DTMF - fire up wireshark (or Tshark) and capture the trace. You can see the RTP DTMF in the trace and hear the clamped audio from the carrier perhaps. Or just hear the inband tones. It gives a good visual of what is going on.


  On Thu, May 6, 2010 at 1:25 PM, Madovsky <infos at madovsky.org> wrote:

    I anyone knows a link of a programming example of how
    to inject rfc2833 in RTP flow (doesn't matter the language)
    It would be very useful

    Thanks

    F
      ----- Original Message ----- 
      From: Madovsky 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Thursday, May 06, 2010 12:39 PM
      Subject: Re: [Freeswitch-users] rfc2833 confusion


      god, so many confusion for only an audio tone ! ;)
      anyway if I understand it's called inband by the fact that it's in RTP flow onlly.
      so the rfc2833 packet is only an event payload ...

        ----- Original Message ----- 
        From: Anthony Minessale 
        To: freeswitch-users at lists.freeswitch.org 
        Sent: Thursday, May 06, 2010 11:52 AM
        Subject: Re: [Freeswitch-users] rfc2833 confusion


        both 2833 and actual tones in the audio stream are often called in-band because the 2833 is just a transport for the inband tones that is assumed to be reconstructed on the other side.  Some people refer to inband as the real tones and 2833 as not inband because it's not the tones.  So the one you have to look out for is people who call 2833 inband and do not clarify that it's inband packetized according the RFC2833  


        The one that is not inband for sure is INFO (which has been dropped from the standard in favor of some ridiculous SUBSCRIBE based xml exchange) 


        Like all things in SIP and anything else that comes with an RFC even if you know the answer you can't rely on trusting that answer because the real world will do whatever it wants and invalidate you. =D




        On Thu, May 6, 2010 at 10:36 AM, Madovsky <infos at madovsky.org> wrote:

          HI,

          is rfc2833 inband or outband ? is it only event or mixed event and sound ?
          I'm confused after read articles on this subject

          Thx

          F

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





        -- 
        Anthony Minessale II

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

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

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



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


        _______________________________________________
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        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From infos at madovsky.org  Thu May  6 11:15:48 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 6 May 2010 14:15:48 -0400
Subject: [Freeswitch-users] Fw:  rfc2833 confusion
Message-ID: <B270322143E74F04A9EEF3359B31EB46@MOBILEE1705>

is this the right link ?

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


----- Original Message ----- 
From: Madovsky 
To: freeswitch-users at lists.freeswitch.org 
Sent: Thursday, May 06, 2010 2:10 PM
Subject: Re: [Freeswitch-users] rfc2833 confusion


But how to catch digits form a dialplan when the call
is already on ?
  ----- Original Message ----- 
  From: Phillip Jones 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, May 06, 2010 1:43 PM
  Subject: Re: [Freeswitch-users] rfc2833 confusion


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

  Don't get to hung up on the inband / out of band thing. This will be negotiated by FS and the carrier when the call is set up. For instance, when you use G729 you need out of band DTMF but for G711 you can use both. This might change from call you call, depending on what media gateways can handle.

  If you want to "see" and hear DTMF - fire up wireshark (or Tshark) and capture the trace. You can see the RTP DTMF in the trace and hear the clamped audio from the carrier perhaps. Or just hear the inband tones. It gives a good visual of what is going on.


  On Thu, May 6, 2010 at 1:25 PM, Madovsky <infos at madovsky.org> wrote:

    I anyone knows a link of a programming example of how
    to inject rfc2833 in RTP flow (doesn't matter the language)
    It would be very useful

    Thanks

    F
      ----- Original Message ----- 
      From: Madovsky 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Thursday, May 06, 2010 12:39 PM
      Subject: Re: [Freeswitch-users] rfc2833 confusion


      god, so many confusion for only an audio tone ! ;)
      anyway if I understand it's called inband by the fact that it's in RTP flow onlly.
      so the rfc2833 packet is only an event payload ...

        ----- Original Message ----- 
        From: Anthony Minessale 
        To: freeswitch-users at lists.freeswitch.org 
        Sent: Thursday, May 06, 2010 11:52 AM
        Subject: Re: [Freeswitch-users] rfc2833 confusion


        both 2833 and actual tones in the audio stream are often called in-band because the 2833 is just a transport for the inband tones that is assumed to be reconstructed on the other side.  Some people refer to inband as the real tones and 2833 as not inband because it's not the tones.  So the one you have to look out for is people who call 2833 inband and do not clarify that it's inband packetized according the RFC2833  


        The one that is not inband for sure is INFO (which has been dropped from the standard in favor of some ridiculous SUBSCRIBE based xml exchange) 


        Like all things in SIP and anything else that comes with an RFC even if you know the answer you can't rely on trusting that answer because the real world will do whatever it wants and invalidate you. =D




        On Thu, May 6, 2010 at 10:36 AM, Madovsky <infos at madovsky.org> wrote:

          HI,

          is rfc2833 inband or outband ? is it only event or mixed event and sound ?
          I'm confused after read articles on this subject

          Thx

          F

          _______________________________________________
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          FreeSWITCH-users at lists.freeswitch.org
          http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
          UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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        -- 
        Anthony Minessale II

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

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

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



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


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        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From robert.hadley at teotech.com  Thu May  6 11:59:16 2010
From: robert.hadley at teotech.com (Robert Hadley)
Date: Thu, 6 May 2010 11:59:16 -0700
Subject: [Freeswitch-users] How to turn off SW DTMF detect (for DTMF talkoff
	issue)
Message-ID: <5418FA83F60147DFAA7D289D13357070@greyhawk.tonecommander.com>

 

Our Freeswitch implementation has two second audio dropouts on calls that
pass through the Sangoma PRI card caused by DTMF talkoff.

 

We know how to turn HW DTMF detection off, but then FS automatically enables
software DTMF detect.  How do we prevent SW DTMF detection in FS?

 

Our attempt to manage when HW DTMF detection is active is not very clean
(see pastebin report of things we tried at
http://pastebin.freeswitch.org/12923).

 

I tried using the start_dtmf and stop_dtmf applications in the dialplan but
they didn't work to prevent talkoff or splits, and the stop_dtmf application
stopped audio for internal calls.  Is there a better way to control HW DTMF
detection or DTMF line splits from the dialplan?

 

Thanks,

Bob

Robert

 

 

 

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From pjintheusa at gmail.com  Thu May  6 12:06:08 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Thu, 6 May 2010 15:06:08 -0400
Subject: [Freeswitch-users] Fw: rfc2833 confusion
In-Reply-To: <B270322143E74F04A9EEF3359B31EB46@MOBILEE1705>
References: <B270322143E74F04A9EEF3359B31EB46@MOBILEE1705>
Message-ID: <p2h367751821005061206w8eb9c3b1o5e5c25385048e822@mail.gmail.com>

Yes that allows you to play a prompt and then wait for the person on the
line to enter digits.

On Thu, May 6, 2010 at 2:15 PM, Madovsky <infos at madovsky.org> wrote:

>  is this the right link ?
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits
>
>
> ----- Original Message -----
> *From:* Madovsky <infos at madovsky.org>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Thursday, May 06, 2010 2:10 PM
> *Subject:* Re: [Freeswitch-users] rfc2833 confusion
>
> But how to catch digits form a dialplan when the call
> is already on ?
>
> ----- Original Message -----
> *From:* Phillip Jones <pjintheusa at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Thursday, May 06, 2010 1:43 PM
> *Subject:* Re: [Freeswitch-users] rfc2833 confusion
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_send_dtmf
>
> Don't get to hung up on the inband / out of band thing. This will be
> negotiated by FS and the carrier when the call is set up. For instance, when
> you use G729 you need out of band DTMF but for G711 you can use both. This
> might change from call you call, depending on what media gateways can
> handle.
>
> If you want to "see" and hear DTMF - fire up wireshark (or Tshark) and
> capture the trace. You can see the RTP DTMF in the trace and hear the
> clamped audio from the carrier perhaps. Or just hear the inband tones. It
> gives a good visual of what is going on.
>
> On Thu, May 6, 2010 at 1:25 PM, Madovsky <infos at madovsky.org> wrote:
>
>>  I anyone knows a link of a programming example of how
>> to inject rfc2833 in RTP flow (doesn't matter the language)
>> It would be very useful
>>
>> Thanks
>>
>> F
>>
>>  ----- Original Message -----
>> *From:* Madovsky <infos at madovsky.org>
>> *To:* freeswitch-users at lists.freeswitch.org
>>   *Sent:* Thursday, May 06, 2010 12:39 PM
>> *Subject:* Re: [Freeswitch-users] rfc2833 confusion
>>
>> god, so many confusion for only an audio tone ! ;)
>> anyway if I understand it's called inband by the fact that it's in RTP
>> flow onlly.
>> so the rfc2833 packet is only an event payload ...
>>
>>
>> ----- Original Message -----
>> *From:* Anthony Minessale <anthony.minessale at gmail.com>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Sent:* Thursday, May 06, 2010 11:52 AM
>> *Subject:* Re: [Freeswitch-users] rfc2833 confusion
>>
>> both 2833 and actual tones in the audio stream are often called in-band
>> because the 2833 is just a transport for the inband tones that is assumed to
>> be reconstructed on the other side.  Some people refer to inband as the real
>> tones and 2833 as not inband because it's not the tones.  So the one you
>> have to look out for is people who call 2833 inband and do not clarify that
>> it's inband packetized according the RFC2833
>>
>> The one that is not inband for sure is INFO (which has been dropped from
>> the standard in favor of some ridiculous SUBSCRIBE based xml exchange)
>>
>> Like all things in SIP and anything else that comes with an RFC even if
>> you know the answer you can't rely on trusting that answer because the real
>> world will do whatever it wants and invalidate you. =D
>>
>>
>> On Thu, May 6, 2010 at 10:36 AM, Madovsky <infos at madovsky.org> wrote:
>>
>>>  HI,
>>>
>>> is rfc2833 inband or outband ? is it only event or mixed event and sound
>>> ?
>>> I'm confused after read articles on this subject
>>>
>>> Thx
>>>
>>> F
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <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-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>
>
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From infos at madovsky.org  Thu May  6 12:23:04 2010
From: infos at madovsky.org (Madovsky)
Date: Thu, 6 May 2010 15:23:04 -0400
Subject: [Freeswitch-users] Fw: rfc2833 confusion
References: <B270322143E74F04A9EEF3359B31EB46@MOBILEE1705>
	<p2h367751821005061206w8eb9c3b1o5e5c25385048e822@mail.gmail.com>
Message-ID: <247242E3584D4DBA852DFF721339D817@MOBILEE1705>

right,
but in case of the SIP caller needs to send digits to FS,
his SIP phone should in any way send DTMF, isn't it ?
in fact I'm trying to implement rfc2833 in my own sip phone through RTP,
but if there is a way to send from caller a string var representing the digits typed to FS once bridge is done
and transfer it as dtmf digits I will be happy too... 

  ----- Original Message ----- 
  From: Phillip Jones 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Thursday, May 06, 2010 3:06 PM
  Subject: Re: [Freeswitch-users] Fw: rfc2833 confusion


  Yes that allows you to play a prompt and then wait for the person on the line to enter digits.


  On Thu, May 6, 2010 at 2:15 PM, Madovsky <infos at madovsky.org> wrote:

    is this the right link ?

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


    ----- Original Message ----- 
    From: Madovsky 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Thursday, May 06, 2010 2:10 PM
    Subject: Re: [Freeswitch-users] rfc2833 confusion


    But how to catch digits form a dialplan when the call
    is already on ?
      ----- Original Message ----- 
      From: Phillip Jones 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Thursday, May 06, 2010 1:43 PM
      Subject: Re: [Freeswitch-users] rfc2833 confusion


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

      Don't get to hung up on the inband / out of band thing. This will be negotiated by FS and the carrier when the call is set up. For instance, when you use G729 you need out of band DTMF but for G711 you can use both. This might change from call you call, depending on what media gateways can handle.

      If you want to "see" and hear DTMF - fire up wireshark (or Tshark) and capture the trace. You can see the RTP DTMF in the trace and hear the clamped audio from the carrier perhaps. Or just hear the inband tones. It gives a good visual of what is going on.


      On Thu, May 6, 2010 at 1:25 PM, Madovsky <infos at madovsky.org> wrote:

        I anyone knows a link of a programming example of how
        to inject rfc2833 in RTP flow (doesn't matter the language)
        It would be very useful

        Thanks

        F
          ----- Original Message ----- 
          From: Madovsky 
          To: freeswitch-users at lists.freeswitch.org 
          Sent: Thursday, May 06, 2010 12:39 PM
          Subject: Re: [Freeswitch-users] rfc2833 confusion


          god, so many confusion for only an audio tone ! ;)
          anyway if I understand it's called inband by the fact that it's in RTP flow onlly.
          so the rfc2833 packet is only an event payload ...

            ----- Original Message ----- 
            From: Anthony Minessale 
            To: freeswitch-users at lists.freeswitch.org 
            Sent: Thursday, May 06, 2010 11:52 AM
            Subject: Re: [Freeswitch-users] rfc2833 confusion


            both 2833 and actual tones in the audio stream are often called in-band because the 2833 is just a transport for the inband tones that is assumed to be reconstructed on the other side.  Some people refer to inband as the real tones and 2833 as not inband because it's not the tones.  So the one you have to look out for is people who call 2833 inband and do not clarify that it's inband packetized according the RFC2833  


            The one that is not inband for sure is INFO (which has been dropped from the standard in favor of some ridiculous SUBSCRIBE based xml exchange) 


            Like all things in SIP and anything else that comes with an RFC even if you know the answer you can't rely on trusting that answer because the real world will do whatever it wants and invalidate you. =D




            On Thu, May 6, 2010 at 10:36 AM, Madovsky <infos at madovsky.org> wrote:

              HI,

              is rfc2833 inband or outband ? is it only event or mixed event and sound ?
              I'm confused after read articles on this subject

              Thx

              F

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





            -- 
            Anthony Minessale II

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

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

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



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


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            UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From max.clark at gmail.com  Thu May  6 15:41:15 2010
From: max.clark at gmail.com (Max Clark)
Date: Thu, 6 May 2010 15:41:15 -0700
Subject: [Freeswitch-users] 4x T1 PRI Gateway
In-Reply-To: <SNT135-w1218CD42C8476DCF66433FD2F50@phx.gbl>
References: <p2s2fa1e1781005060736u41f4d0d9w1efa59f8a28172fa@mail.gmail.com>
	<SNT135-w1218CD42C8476DCF66433FD2F50@phx.gbl>
Message-ID: <v2x2fa1e1781005061541qf523da1cu168973619a54ac05@mail.gmail.com>

If I alter my requirement a little and say that I want to purchase an
appliance to give to a non-technical end user that I will not have to
support long term what's my best bet?

Audiocodes seems incredibly expensive - is it worth it? Cisco 5350
seems to be easily found on the secondary market. Is the
configuration/feature set usable? I am investigating Aculab which
seems interesting - anything else I should look at?

Thanks,
Max

On Thu, May 6, 2010 at 8:49 AM, Jan Berger <janvb at live.com> wrote:
> What signalling protocol will you?use to connect to the legacy PBX and how
> much work are you prepard to do yourself? It's critical that you get exact
> information on your legacy PBX signalling
>
> - R1
> - R2
> - DMS
> - AT&T
> - ANSI ISDN
> - SS7
> etc...
>
> In general - FreeSWITCH can do it with a medium PC and a 4 x T1 card from
> Sangoma etc. The cost for a 4xT1 solution will be significantly cheaper than
> anything than a so-called plug & play box.
>
> Jan
>
>
>> Date: Thu, 6 May 2010 07:36:04 -0700
>> From: max.clark at gmail.com
>> To: Freeswitch-users at lists.freeswitch.org
>> Subject: [Freeswitch-users] 4x T1 PRI Gateway
>>
>> Hello,
>>
>> I am looking for a 4 port T1/PRI to SIP gateway to interconnect a
>> legacy PBX to a SIP trunk. After spending some time with Google I have
>> come up with the following list of devices that fit the required
>> number of T1 interfaces:
>>
>> Dialogic DMG2120DTI
>> Cisco AS5350
>> Quintum Tenor DX
>> Patton SmartNode 4960
>> Audiocodes Mediant 1000/2000
>>
>> Any opinions, personal experiences, advice, or additional devices I
>> should be looking at would be greatly appreciated.
>>
>> Best,
>> Max
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> ________________________________
> Hotmail: Powerful Free email with security by Microsoft. Get it now.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From casteven at gmail.com  Thu May  6 15:51:16 2010
From: casteven at gmail.com (Campbell Steven)
Date: Fri, 7 May 2010 10:51:16 +1200
Subject: [Freeswitch-users] 4x T1 PRI Gateway
In-Reply-To: <v2x2fa1e1781005061541qf523da1cu168973619a54ac05@mail.gmail.com>
References: <p2s2fa1e1781005060736u41f4d0d9w1efa59f8a28172fa@mail.gmail.com>
	<SNT135-w1218CD42C8476DCF66433FD2F50@phx.gbl>
	<v2x2fa1e1781005061541qf523da1cu168973619a54ac05@mail.gmail.com>
Message-ID: <v2u6a31a78a1005061551g4d5347c6ua7921c5c2f8e7eb5@mail.gmail.com>

I've had good experiences with the Patton SN4960 in this situation.

Campbell

On Fri, May 7, 2010 at 10:41 AM, Max Clark <max.clark at gmail.com> wrote:
> If I alter my requirement a little and say that I want to purchase an
> appliance to give to a non-technical end user that I will not have to
> support long term what's my best bet?
>
> Audiocodes seems incredibly expensive - is it worth it? Cisco 5350
> seems to be easily found on the secondary market. Is the
> configuration/feature set usable? I am investigating Aculab which
> seems interesting - anything else I should look at?
>
> Thanks,
> Max
>
> On Thu, May 6, 2010 at 8:49 AM, Jan Berger <janvb at live.com> wrote:
>> What signalling protocol will you?use to connect to the legacy PBX and how
>> much work are you prepard to do yourself? It's critical that you get exact
>> information on your legacy PBX signalling
>>
>> - R1
>> - R2
>> - DMS
>> - AT&T
>> - ANSI ISDN
>> - SS7
>> etc...
>>
>> In general - FreeSWITCH can do it with a medium PC and a 4 x T1 card from
>> Sangoma etc. The cost for a 4xT1 solution will be significantly cheaper than
>> anything than a so-called plug & play box.
>>
>> Jan
>>
>>
>>> Date: Thu, 6 May 2010 07:36:04 -0700
>>> From: max.clark at gmail.com
>>> To: Freeswitch-users at lists.freeswitch.org
>>> Subject: [Freeswitch-users] 4x T1 PRI Gateway
>>>
>>> Hello,
>>>
>>> I am looking for a 4 port T1/PRI to SIP gateway to interconnect a
>>> legacy PBX to a SIP trunk. After spending some time with Google I have
>>> come up with the following list of devices that fit the required
>>> number of T1 interfaces:
>>>
>>> Dialogic DMG2120DTI
>>> Cisco AS5350
>>> Quintum Tenor DX
>>> Patton SmartNode 4960
>>> Audiocodes Mediant 1000/2000
>>>
>>> Any opinions, personal experiences, advice, or additional devices I
>>> should be looking at would be greatly appreciated.
>>>
>>> Best,
>>> Max
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> ________________________________
>> Hotmail: Powerful Free email with security by Microsoft. Get it now.
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From krice at freeswitch.org  Thu May  6 15:56:14 2010
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 06 May 2010 17:56:14 -0500
Subject: [Freeswitch-users] 4x T1 PRI Gateway
In-Reply-To: <v2x2fa1e1781005061541qf523da1cu168973619a54ac05@mail.gmail.com>
Message-ID: <C808B33E.6CB03%krice@freeswitch.org>

AS5300 series are great T1 gateways. If you are purchasing off of ebay, make
sure you get one tagged "VoIP" there are 2 different cards that can go in
these things one is MICA modems which will do you no good. The other is the
DSP cards which is what you want. You can barely give away MICA Modems these
days (I have several if anyone wants them you pay shipping lol)

K


On 5/6/10 5:41 PM, "Max Clark" <max.clark at gmail.com> wrote:

> If I alter my requirement a little and say that I want to purchase an
> appliance to give to a non-technical end user that I will not have to
> support long term what's my best bet?
> 
> Audiocodes seems incredibly expensive - is it worth it? Cisco 5350
> seems to be easily found on the secondary market. Is the
> configuration/feature set usable? I am investigating Aculab which
> seems interesting - anything else I should look at?
> 
> Thanks,
> Max
> 
> On Thu, May 6, 2010 at 8:49 AM, Jan Berger <janvb at live.com> wrote:
>> What signalling protocol will you?use to connect to the legacy PBX and how
>> much work are you prepard to do yourself? It's critical that you get exact
>> information on your legacy PBX signalling
>> 
>> - R1
>> - R2
>> - DMS
>> - AT&T
>> - ANSI ISDN
>> - SS7
>> etc...
>> 
>> In general - FreeSWITCH can do it with a medium PC and a 4 x T1 card from
>> Sangoma etc. The cost for a 4xT1 solution will be significantly cheaper than
>> anything than a so-called plug & play box.
>> 
>> Jan
>> 
>> 
>>> Date: Thu, 6 May 2010 07:36:04 -0700
>>> From: max.clark at gmail.com
>>> To: Freeswitch-users at lists.freeswitch.org
>>> Subject: [Freeswitch-users] 4x T1 PRI Gateway
>>> 
>>> Hello,
>>> 
>>> I am looking for a 4 port T1/PRI to SIP gateway to interconnect a
>>> legacy PBX to a SIP trunk. After spending some time with Google I have
>>> come up with the following list of devices that fit the required
>>> number of T1 interfaces:
>>> 
>>> Dialogic DMG2120DTI
>>> Cisco AS5350
>>> Quintum Tenor DX
>>> Patton SmartNode 4960
>>> Audiocodes Mediant 1000/2000
>>> 
>>> Any opinions, personal experiences, advice, or additional devices I
>>> should be looking at would be greatly appreciated.
>>> 
>>> Best,
>>> Max
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> ________________________________
>> Hotmail: Powerful Free email with security by Microsoft. Get it now.
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From anthony.minessale at gmail.com  Thu May  6 16:17:25 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 6 May 2010 18:17:25 -0500
Subject: [Freeswitch-users] 4x T1 PRI Gateway
In-Reply-To: <C808B33E.6CB03%krice@freeswitch.org>
References: <v2x2fa1e1781005061541qf523da1cu168973619a54ac05@mail.gmail.com>
	<C808B33E.6CB03%krice@freeswitch.org>
Message-ID: <n2i191c3a031005061617z89422426xcdc04da4a9d99d30@mail.gmail.com>

Ken Rice?
Where the heck have you been?


On Thu, May 6, 2010 at 5:56 PM, Ken Rice <krice at freeswitch.org> wrote:

> AS5300 series are great T1 gateways. If you are purchasing off of ebay,
> make
> sure you get one tagged "VoIP" there are 2 different cards that can go in
> these things one is MICA modems which will do you no good. The other is the
> DSP cards which is what you want. You can barely give away MICA Modems
> these
> days (I have several if anyone wants them you pay shipping lol)
>
> K
>
>
> On 5/6/10 5:41 PM, "Max Clark" <max.clark at gmail.com> wrote:
>
> > If I alter my requirement a little and say that I want to purchase an
> > appliance to give to a non-technical end user that I will not have to
> > support long term what's my best bet?
> >
> > Audiocodes seems incredibly expensive - is it worth it? Cisco 5350
> > seems to be easily found on the secondary market. Is the
> > configuration/feature set usable? I am investigating Aculab which
> > seems interesting - anything else I should look at?
> >
> > Thanks,
> > Max
> >
> > On Thu, May 6, 2010 at 8:49 AM, Jan Berger <janvb at live.com> wrote:
> >> What signalling protocol will you use to connect to the legacy PBX and
> how
> >> much work are you prepard to do yourself? It's critical that you get
> exact
> >> information on your legacy PBX signalling
> >>
> >> - R1
> >> - R2
> >> - DMS
> >> - AT&T
> >> - ANSI ISDN
> >> - SS7
> >> etc...
> >>
> >> In general - FreeSWITCH can do it with a medium PC and a 4 x T1 card
> from
> >> Sangoma etc. The cost for a 4xT1 solution will be significantly cheaper
> than
> >> anything than a so-called plug & play box.
> >>
> >> Jan
> >>
> >>
> >>> Date: Thu, 6 May 2010 07:36:04 -0700
> >>> From: max.clark at gmail.com
> >>> To: Freeswitch-users at lists.freeswitch.org
> >>> Subject: [Freeswitch-users] 4x T1 PRI Gateway
> >>>
> >>> Hello,
> >>>
> >>> I am looking for a 4 port T1/PRI to SIP gateway to interconnect a
> >>> legacy PBX to a SIP trunk. After spending some time with Google I have
> >>> come up with the following list of devices that fit the required
> >>> number of T1 interfaces:
> >>>
> >>> Dialogic DMG2120DTI
> >>> Cisco AS5350
> >>> Quintum Tenor DX
> >>> Patton SmartNode 4960
> >>> Audiocodes Mediant 1000/2000
> >>>
> >>> Any opinions, personal experiences, advice, or additional devices I
> >>> should be looking at would be greatly appreciated.
> >>>
> >>> Best,
> >>> Max
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >> ________________________________
> >> Hotmail: Powerful Free email with security by Microsoft. Get it now.
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
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
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From jmesquita at freeswitch.org  Thu May  6 16:24:30 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Thu, 6 May 2010 20:24:30 -0300
Subject: [Freeswitch-users] 4x T1 PRI Gateway
In-Reply-To: <n2i191c3a031005061617z89422426xcdc04da4a9d99d30@mail.gmail.com>
References: <v2x2fa1e1781005061541qf523da1cu168973619a54ac05@mail.gmail.com>
	<C808B33E.6CB03%krice@freeswitch.org>
	<n2i191c3a031005061617z89422426xcdc04da4a9d99d30@mail.gmail.com>
Message-ID: <g2rebbc2d551005061624k1c3e9b2ak47e2154c0c8882eb@mail.gmail.com>

Rising from the shadows like phoenix...
JM


On Thu, May 6, 2010 at 8:17 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Ken Rice?
> Where the heck have you been?
>
>
> On Thu, May 6, 2010 at 5:56 PM, Ken Rice <krice at freeswitch.org> wrote:
>
>> AS5300 series are great T1 gateways. If you are purchasing off of ebay,
>> make
>> sure you get one tagged "VoIP" there are 2 different cards that can go in
>> these things one is MICA modems which will do you no good. The other is
>> the
>> DSP cards which is what you want. You can barely give away MICA Modems
>> these
>> days (I have several if anyone wants them you pay shipping lol)
>>
>> K
>>
>>
>> On 5/6/10 5:41 PM, "Max Clark" <max.clark at gmail.com> wrote:
>>
>> > If I alter my requirement a little and say that I want to purchase an
>> > appliance to give to a non-technical end user that I will not have to
>> > support long term what's my best bet?
>> >
>> > Audiocodes seems incredibly expensive - is it worth it? Cisco 5350
>> > seems to be easily found on the secondary market. Is the
>> > configuration/feature set usable? I am investigating Aculab which
>> > seems interesting - anything else I should look at?
>> >
>> > Thanks,
>> > Max
>> >
>> > On Thu, May 6, 2010 at 8:49 AM, Jan Berger <janvb at live.com> wrote:
>> >> What signalling protocol will you use to connect to the legacy PBX and
>> how
>> >> much work are you prepard to do yourself? It's critical that you get
>> exact
>> >> information on your legacy PBX signalling
>> >>
>> >> - R1
>> >> - R2
>> >> - DMS
>> >> - AT&T
>> >> - ANSI ISDN
>> >> - SS7
>> >> etc...
>> >>
>> >> In general - FreeSWITCH can do it with a medium PC and a 4 x T1 card
>> from
>> >> Sangoma etc. The cost for a 4xT1 solution will be significantly cheaper
>> than
>> >> anything than a so-called plug & play box.
>> >>
>> >> Jan
>> >>
>> >>
>> >>> Date: Thu, 6 May 2010 07:36:04 -0700
>> >>> From: max.clark at gmail.com
>> >>> To: Freeswitch-users at lists.freeswitch.org
>> >>> Subject: [Freeswitch-users] 4x T1 PRI Gateway
>> >>>
>> >>> Hello,
>> >>>
>> >>> I am looking for a 4 port T1/PRI to SIP gateway to interconnect a
>> >>> legacy PBX to a SIP trunk. After spending some time with Google I have
>> >>> come up with the following list of devices that fit the required
>> >>> number of T1 interfaces:
>> >>>
>> >>> Dialogic DMG2120DTI
>> >>> Cisco AS5350
>> >>> Quintum Tenor DX
>> >>> Patton SmartNode 4960
>> >>> Audiocodes Mediant 1000/2000
>> >>>
>> >>> Any opinions, personal experiences, advice, or additional devices I
>> >>> should be looking at would be greatly appreciated.
>> >>>
>> >>> Best,
>> >>> Max
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >> ________________________________
>> >> Hotmail: Powerful Free email with security by Microsoft. Get it now.
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jaybinks at gmail.com  Thu May  6 16:42:38 2010
From: jaybinks at gmail.com (jay binks)
Date: Fri, 7 May 2010 09:42:38 +1000
Subject: [Freeswitch-users] 4x T1 PRI Gateway
In-Reply-To: <n2i191c3a031005061617z89422426xcdc04da4a9d99d30@mail.gmail.com>
References: <v2x2fa1e1781005061541qf523da1cu168973619a54ac05@mail.gmail.com> 
	<C808B33E.6CB03%krice@freeswitch.org>
	<n2i191c3a031005061617z89422426xcdc04da4a9d99d30@mail.gmail.com>
Message-ID: <x2sec2ce17e1005061642t7ca6477chbe098765aa23a8a9@mail.gmail.com>

yo ken, is your IM / IRC client broken ??
:)

oh yea... back on topic..
Audiocodes Mediant 2k's can go for reasonable prices on ebay.
worth a look...

J

On Fri, May 7, 2010 at 9:17 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Ken Rice?
> Where the heck have you been?
>
>
> On Thu, May 6, 2010 at 5:56 PM, Ken Rice <krice at freeswitch.org> wrote:
>
>> AS5300 series are great T1 gateways. If you are purchasing off of ebay,
>> make
>> sure you get one tagged "VoIP" there are 2 different cards that can go in
>> these things one is MICA modems which will do you no good. The other is
>> the
>> DSP cards which is what you want. You can barely give away MICA Modems
>> these
>> days (I have several if anyone wants them you pay shipping lol)
>>
>> K
>>
>>
>> On 5/6/10 5:41 PM, "Max Clark" <max.clark at gmail.com> wrote:
>>
>> > If I alter my requirement a little and say that I want to purchase an
>> > appliance to give to a non-technical end user that I will not have to
>> > support long term what's my best bet?
>> >
>> > Audiocodes seems incredibly expensive - is it worth it? Cisco 5350
>> > seems to be easily found on the secondary market. Is the
>> > configuration/feature set usable? I am investigating Aculab which
>> > seems interesting - anything else I should look at?
>> >
>> > Thanks,
>> > Max
>> >
>> > On Thu, May 6, 2010 at 8:49 AM, Jan Berger <janvb at live.com> wrote:
>> >> What signalling protocol will you use to connect to the legacy PBX and
>> how
>> >> much work are you prepard to do yourself? It's critical that you get
>> exact
>> >> information on your legacy PBX signalling
>> >>
>> >> - R1
>> >> - R2
>> >> - DMS
>> >> - AT&T
>> >> - ANSI ISDN
>> >> - SS7
>> >> etc...
>> >>
>> >> In general - FreeSWITCH can do it with a medium PC and a 4 x T1 card
>> from
>> >> Sangoma etc. The cost for a 4xT1 solution will be significantly cheaper
>> than
>> >> anything than a so-called plug & play box.
>> >>
>> >> Jan
>> >>
>> >>
>> >>> Date: Thu, 6 May 2010 07:36:04 -0700
>> >>> From: max.clark at gmail.com
>> >>> To: Freeswitch-users at lists.freeswitch.org
>> >>> Subject: [Freeswitch-users] 4x T1 PRI Gateway
>> >>>
>> >>> Hello,
>> >>>
>> >>> I am looking for a 4 port T1/PRI to SIP gateway to interconnect a
>> >>> legacy PBX to a SIP trunk. After spending some time with Google I have
>> >>> come up with the following list of devices that fit the required
>> >>> number of T1 interfaces:
>> >>>
>> >>> Dialogic DMG2120DTI
>> >>> Cisco AS5350
>> >>> Quintum Tenor DX
>> >>> Patton SmartNode 4960
>> >>> Audiocodes Mediant 1000/2000
>> >>>
>> >>> Any opinions, personal experiences, advice, or additional devices I
>> >>> should be looking at would be greatly appreciated.
>> >>>
>> >>> Best,
>> >>> Max
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >> ________________________________
>> >> Hotmail: Powerful Free email with security by Microsoft. Get it now.
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Sincerely

Jay
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From lfurrea at gmail.com  Thu May  6 16:53:36 2010
From: lfurrea at gmail.com (Luis F Urrea)
Date: Thu, 6 May 2010 18:53:36 -0500
Subject: [Freeswitch-users] 4x T1 PRI Gateway
In-Reply-To: <g2rebbc2d551005061624k1c3e9b2ak47e2154c0c8882eb@mail.gmail.com>
References: <v2x2fa1e1781005061541qf523da1cu168973619a54ac05@mail.gmail.com>
	<C808B33E.6CB03%krice@freeswitch.org>
	<n2i191c3a031005061617z89422426xcdc04da4a9d99d30@mail.gmail.com>
	<g2rebbc2d551005061624k1c3e9b2ak47e2154c0c8882eb@mail.gmail.com>
Message-ID: <n2lf120f4c81005061653o55cd11ddx1ae940a80b444586@mail.gmail.com>

I do not know about digital audiocodes but I had horrible experiences with
analog Audiocodes gateways. The crappy device every once in a while simply
looses IP connectivity and it was painful to get it back, I even had to
reset to factory defaults several times . Consider the fact that you may be
miles away from the site and you will understand why I find them painful.

Audiocodes support is also crap. you can only try to get support through
your reseller.

+1 for Patton 4960

Smartware firmware is really powerful and stable with more than 8 years of
development. CLI is nice and powerful as well,  debugging through CLI can
help a lot. I don't like the Web interface and find it confusing but GUIs
never appealed to me :p

Smartware documentation is good as well.

At least for analog gateways SNMP MIBs are alright but not very helpful to
monitor every aspect you can think of, but I think the 4960 may include more
than the analog ones, you can also find documentation on available MIBs.

You can get support directly from Patton.

I have several analog ones in production and never had to deal with them for
over a year.

My 2 cents and humble opinion.

Hope it helps
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From bwibowo at gmail.com  Thu May  6 16:42:52 2010
From: bwibowo at gmail.com (budi wibowo)
Date: Fri, 7 May 2010 06:42:52 +0700
Subject: [Freeswitch-users] outgoing gateway
Message-ID: <l2o547ec34a1005061642y37cd9880tac6e3aef1da1be1@mail.gmail.com>

dear all
i'm new to freeswitch, and i found difficulties to connect fs to my mera
switch.
here's the topology

extension-------------->FS-------------> meraswitch----------------->pstn

what need to be configured to connect FS to meraswitch that run SIP protocol

TIA

budi wibowo
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From mike at jerris.com  Thu May  6 17:16:41 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 6 May 2010 20:16:41 -0400
Subject: [Freeswitch-users] Encoding issue with mod_xml_cdr (apparently)
In-Reply-To: <20100504071146.GA27482@econcept04.econcept.es>
References: <20100504071146.GA27482@econcept04.econcept.es>
Message-ID: <26665FA4-2A27-49EE-A306-3DD19B7B1333@jerris.com>

does a packet trace of the post request confirm this?  If it does, please open a bug on http://jira.freeswitch.org about this issue.

Mike

On May 4, 2010, at 3:11 AM, Isaac Jurado wrote:

> Hi,
> 
> We are setting up a web service for mod_xml_cdr but we have problems
> with non-ascii characters encoded in UTF-8.  The POST request
> (url-encoded) stops its data when it finds the first non-ascii, like the
> following:
> 
> -----8<-----
> <?xml version="1.0"?>
> <cdr>
>  <channel_data>
>    <state>CS_REPORTING</state>
>    <direction>inbound</direction>
>    <state_number>11</state_number>
>    <flags>0=1;18=1;36=1;38=1;51=1</flags>
>    <caps>1=1;2=1;3=1</caps>
>  </channel_data>
>  <variables>
>    <uuid>faa6c7e9-9b4a-4ac0-8ebb-971b2e2f8eaa</uuid>
>    <sip_local_network_addr>192.168.1.99</sip_local_network_addr>
>    <sip_network_ip>192.168.1.99</sip_network_ip>
>    <sip_network_port>1024</sip_network_port>
>    <sip_received_ip>192.168.1.99</sip_received_ip>
>    <sip_received_port>1024</sip_received_port>
>    <sip_via_protocol>udp</sip_via_protocol>
>    <sip_authorized>true</sip_authorized>
>    <sip_number_alias>33</sip_number_alias>
>    <sip_auth_username>33</sip_auth_username>
>    <sip_auth_realm>econcept.es</sip_auth_realm>
>    <number_alias>33</number_alias>
>    <user_name>33</user_name>
>    <domain_name>econcept.es</domain_name>
>    <accountcode>33</accountcode>
>    <user_context>econcept.es</user_context>
>    <caller_id_name>Peque%%
> ----->8-----
> 
> This is what the web service finds under the 'cdr' POST key.  The last
> two percent characters seem to precede the spanish '?'.  The
> caller_id_name is supplied at register time by issuing an UTF-8 encoded
> XML to feed mod_xml_curl.
> 
> I've tried disabling the url-encoding like so (xml_cdr.conf.xml):
> 
>    <param name="encode" value="false"/>
> 
> But then, the web service does not appear to be receiving anything.
> 
> Any ideas?
> 
> Cheers.
> 
> -- 
> Isaac Jurado
> Internet Busines Solutions eConcept
> http://www.econcept.es
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mike at jerris.com  Thu May  6 17:19:02 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 6 May 2010 20:19:02 -0400
Subject: [Freeswitch-users] hanguphook arguments
In-Reply-To: <66EA3166EB339A4489B06286C0876A8A0EB894FA@mailserv.Globalive.local>
References: <66EA3166EB339A4489B06286C0876A8A0EB894FA@mailserv.Globalive.local>
Message-ID: <F6EAFB5F-8292-413D-80E0-CE299BE9DB02@jerris.com>

global var?  you can't change the prototype of the callback fucntion.

On May 5, 2010, at 11:55 AM, Christian Damianidis wrote:

> Looking at http://wiki.freeswitch.org/wiki/Example_Hangup_hook
> I see there are some default arguments, however I want to pass in my own.
>  
> This may be some weak javascript knowledge on my part, but when I do
>  
> session.setHangupHook(on_hangup(myparam))
>  
> It executes the function (obviously as the brackets make it a function call).
>  
> Any tips on how to do this, is it possible?

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From brian at freeswitch.org  Thu May  6 17:20:16 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 6 May 2010 19:20:16 -0500
Subject: [Freeswitch-users] Encoding issue with mod_xml_cdr (apparently)
In-Reply-To: <26665FA4-2A27-49EE-A306-3DD19B7B1333@jerris.com>
References: <20100504071146.GA27482@econcept04.econcept.es>
	<26665FA4-2A27-49EE-A306-3DD19B7B1333@jerris.com>
Message-ID: <1871ECF8-82DD-4DB6-9C52-C274648CDE4B@freeswitch.org>

I just put a patch in for the UTF-8 issue.

/b

On May 6, 2010, at 7:16 PM, Michael Jerris wrote:

> does a packet trace of the post request confirm this?  If it does, please open a bug on http://jira.freeswitch.org about this issue.
> 
> Mike

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From mike at jerris.com  Thu May  6 17:24:31 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 6 May 2010 20:24:31 -0400
Subject: [Freeswitch-users] How to turn off SW DTMF detect (for DTMF
	talkoff issue)
In-Reply-To: <5418FA83F60147DFAA7D289D13357070@greyhawk.tonecommander.com>
References: <5418FA83F60147DFAA7D289D13357070@greyhawk.tonecommander.com>
Message-ID: <EDCE5457-01CA-47E9-B3C9-DC494E04F1A8@jerris.com>

I have never seen these dropouts and we use the dtmf detection without issue.  Try contacting sangoma to see if they have a fix for this, I suspect they do already have this in a newer release.

Mike

On May 6, 2010, at 2:59 PM, Robert Hadley wrote:

>  
> Our Freeswitch implementation has two second audio dropouts on calls that pass through the Sangoma PRI card caused by DTMF talkoff.
>  
> We know how to turn HW DTMF detection off, but then FS automatically enables software DTMF detect.  How do we prevent SW DTMF detection in FS?
>  
> Our attempt to manage when HW DTMF detection is active is not very clean (see pastebin report of things we tried at http://pastebin.freeswitch.org/12923).
>  
> I tried using the start_dtmf and stop_dtmf applications in the dialplan but they didn?t work to prevent talkoff or splits, and the stop_dtmf application stopped audio for internal calls.  Is there a better way to control HW DTMF detection or DTMF line splits from the dialplan?

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From djbinter at gmail.com  Thu May  6 20:34:23 2010
From: djbinter at gmail.com (DJB INTERNATIONAL)
Date: Thu, 6 May 2010 20:34:23 -0700
Subject: [Freeswitch-users] outgoing gateway
In-Reply-To: <l2o547ec34a1005061642y37cd9880tac6e3aef1da1be1@mail.gmail.com>
References: <l2o547ec34a1005061642y37cd9880tac6e3aef1da1be1@mail.gmail.com>
Message-ID: <t2u94f7dfb11005062034wc952a56ev92417d4456cc8b55@mail.gmail.com>

It depends on what version of Mera that you are running.  The older version,
you need to have SIPHIT running; however, the newer one like Mera Pro or
Mera II, they should connect with no problem.

-djbinter


On Thu, May 6, 2010 at 4:42 PM, budi wibowo <bwibowo at gmail.com> wrote:

> dear all
> i'm new to freeswitch, and i found difficulties to connect fs to my mera
> switch.
> here's the topology
>
> extension-------------->FS-------------> meraswitch----------------->pstn
>
> what need to be configured to connect FS to meraswitch that run SIP
> protocol
>
> TIA
>
> budi wibowo
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bwibowo at gmail.com  Thu May  6 20:53:05 2010
From: bwibowo at gmail.com (Budi wibowo)
Date: Fri, 7 May 2010 03:53:05 +0000
Subject: [Freeswitch-users] outgoing gateway
In-Reply-To: <t2u94f7dfb11005062034wc952a56ev92417d4456cc8b55@mail.gmail.com>
References: <l2o547ec34a1005061642y37cd9880tac6e3aef1da1be1@mail.gmail.com><t2u94f7dfb11005062034wc952a56ev92417d4456cc8b55@mail.gmail.com>
Message-ID: <1077559925-1273204385-cardhu_decombobulator_blackberry.rim.net-1446324892-@bda057.bisx.prodap.on.blackberry>

Yes I have siphit installed, I tried to make some changes on dialplan file but call always goes to other server, what should I configure to implement this outgoing call 

Tia
Budi 

-----Original Message-----
From: DJB INTERNATIONAL <djbinter at gmail.com>
Date: Thu, 6 May 2010 20:34:23 
To: <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] outgoing gateway

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




From mike at jerris.com  Thu May  6 22:55:00 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 7 May 2010 01:55:00 -0400
Subject: [Freeswitch-users] outgoing gateway
In-Reply-To: <1077559925-1273204385-cardhu_decombobulator_blackberry.rim.net-1446324892-@bda057.bisx.prodap.on.blackberry>
References: <l2o547ec34a1005061642y37cd9880tac6e3aef1da1be1@mail.gmail.com><t2u94f7dfb11005062034wc952a56ev92417d4456cc8b55@mail.gmail.com>
	<1077559925-1273204385-cardhu_decombobulator_blackberry.rim.net-1446324892-@bda057.bisx.prodap.on.blackberry>
Message-ID: <472BC89E-11F0-49C2-8EA5-23783BC38770@jerris.com>

http://wiki.freeswitch.org/wiki/Dialplan

On May 6, 2010, at 11:53 PM, Budi wibowo wrote:

> Yes I have siphit installed, I tried to make some changes on dialplan file but call always goes to other server, what should I configure to implement this outgoing call 
> 

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From babak.freeswitch at gmail.com  Thu May  6 23:36:24 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Fri, 7 May 2010 11:06:24 +0430
Subject: [Freeswitch-users] recording problem
Message-ID: <n2k1fdf92721005062336g154dd71al2888b4253abfc4e2@mail.gmail.com>

Hi I'm using the code below to dispatch customers to operators in c#.
everything is fine but recording is not taking place(I've tested both the
hang up after bridge false and true).
thanks
public class Answering : IAppPlugin
    {

        ManagedSession Session;
        public void Run(AppContext context)
        {
            Session = context.Session;
            Session.HangupFunction = hangupHook;

            if (. . .)      //no operator available wai
            {
                Session.Answer();
                if (Session.Ready())
                {
                    Session.StreamFile(m_sorryPrompt, -1);
                    Session.SetVariable("sepanta_prompt_ended","true");

                        Session.StreamFile(m_musicOnHold, -1);
                }
            }//now u can talk
            if (Session.Ready())
            {
                string ticket = "1234";
                if (!Session.answered())
                    Session.Answer();
                Session.Say(ticket, "en", "NUMBER", "PRONOUNCED",
"FEMININE");
                Session.Execute("Playback",m_readyPrompt);
                Session.SetVariable("exec_after_bridge_app",
"record_session");
                Session.SetVariable("exec_after_bridge_arg", recPath);

 Session.Execute("bridge",string.Format("sofia/internal/{0}%{1}", op,
Session.GetVariable("domain")));
            }
        }
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From janvb at live.com  Thu May  6 23:46:36 2010
From: janvb at live.com (Jan Berger)
Date: Fri, 7 May 2010 08:46:36 +0200
Subject: [Freeswitch-users] 4x T1 PRI Gateway
In-Reply-To: <v2x2fa1e1781005061541qf523da1cu168973619a54ac05@mail.gmail.com>
References: <p2s2fa1e1781005060736u41f4d0d9w1efa59f8a28172fa@mail.gmail.com>,
	<SNT135-w1218CD42C8476DCF66433FD2F50@phx.gbl>,
	<v2x2fa1e1781005061541qf523da1cu168973619a54ac05@mail.gmail.com>
Message-ID: <SNT135-w11FBF4CCFD6607CD58686DD2F60@phx.gbl>


Most legacy PBX's can actually add SIP trunking in software and upgrade.
 
> Date: Thu, 6 May 2010 15:41:15 -0700
> From: max.clark at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] 4x T1 PRI Gateway
> 
> If I alter my requirement a little and say that I want to purchase an
> appliance to give to a non-technical end user that I will not have to
> support long term what's my best bet?
> 
> Audiocodes seems incredibly expensive - is it worth it? Cisco 5350
> seems to be easily found on the secondary market. Is the
> configuration/feature set usable? I am investigating Aculab which
> seems interesting - anything else I should look at?
> 
> Thanks,
> Max
> 
> On Thu, May 6, 2010 at 8:49 AM, Jan Berger <janvb at live.com> wrote:
> > What signalling protocol will you use to connect to the legacy PBX and how
> > much work are you prepard to do yourself? It's critical that you get exact
> > information on your legacy PBX signalling
> >
> > - R1
> > - R2
> > - DMS
> > - AT&T
> > - ANSI ISDN
> > - SS7
> > etc...
> >
> > In general - FreeSWITCH can do it with a medium PC and a 4 x T1 card from
> > Sangoma etc. The cost for a 4xT1 solution will be significantly cheaper than
> > anything than a so-called plug & play box.
> >
> > Jan
> >
> >
> >> Date: Thu, 6 May 2010 07:36:04 -0700
> >> From: max.clark at gmail.com
> >> To: Freeswitch-users at lists.freeswitch.org
> >> Subject: [Freeswitch-users] 4x T1 PRI Gateway
> >>
> >> Hello,
> >>
> >> I am looking for a 4 port T1/PRI to SIP gateway to interconnect a
> >> legacy PBX to a SIP trunk. After spending some time with Google I have
> >> come up with the following list of devices that fit the required
> >> number of T1 interfaces:
> >>
> >> Dialogic DMG2120DTI
> >> Cisco AS5350
> >> Quintum Tenor DX
> >> Patton SmartNode 4960
> >> Audiocodes Mediant 1000/2000
> >>
> >> Any opinions, personal experiences, advice, or additional devices I
> >> should be looking at would be greatly appreciated.
> >>
> >> Best,
> >> Max
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > ________________________________
> > Hotmail: Powerful Free email with security by Microsoft. Get it now.
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
 		 	   		  
_________________________________________________________________
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From kawarod at laposte.net  Fri May  7 01:25:54 2010
From: kawarod at laposte.net (Rod.)
Date: Fri, 07 May 2010 12:25:54 +0400
Subject: [Freeswitch-users] Registration ODBC feeded by
 another	registrar proxy
In-Reply-To: <w2y367751821005060600k2a49680cybcb92e12fa3c26aa@mail.gmail.com>
References: <4BDED22F.20205@laposte.net>	<C5D5A056-B21C-4343-AA13-95D67B3EADA6@gmail.com>	<4BDF02AE.3010003@laposte.net>	<SNT135-w18CD702457BE5E08B7D51D2F20@phx.gbl>	<4BDFD6FD.2050903@laposte.net>
	<4BE2882A.7080102@laposte.net>
	<w2y367751821005060600k2a49680cybcb92e12fa3c26aa@mail.gmail.com>
Message-ID: <4BE3CE92.20604@laposte.net>

Thanks Philipp,

for the 40 000 users, I'm just playing with FS at the moment. I know 
that clustering is the way to go, but at the moment I'm just trying FS 
as a big PBX.

rod.

Le 06/05/2010 17:00, Phillip Jones a ?crit :
> Ron,
>
> I don't know - but I suspect that expires is a actual time. That would 
> make sense to me - something like:
>
> expires = epochNow + timeout;
>
> As for the 40,000 users - are you really planning doing this on one 
> box? Registrations + there traffic? Surely doing a FS cluster as 
> discussed or having OpenSIPS handle registration would be the way to 
> go here.
>
>
>
> On Thu, May 6, 2010 at 5:13 AM, Rod. <kawarod at laposte.net 
> <mailto:kawarod at laposte.net>> wrote:
>
>     Hi,
>
>     At the moment, I'm registering 40000 users in FS using ODBC with
>     no more than 16 registration/sec.
>     When I do a select in the sip_registrations table, I see that
>     every sip_user have an expires value equal to something like
>     "1273141208" even if I'm using an expire value of 3600 in my sip
>     register request.
>
>     +-----------------------+----------+--------------+----------------+-----------------------------------+-----------------+---------+------------+------------+-------------+---------------+--------------+----------+------------+--------------+--------------+--------------+----------+--------------+------------------+---------------+
>     | call_id               | sip_user | sip_host     | presence_hosts
>     | contact                           | status          | rpid    |
>     expires    | user_agent | server_user | server_host   |
>     profile_name | hostname | network_ip | network_port | sip_username
>     | sip_realm    | mwi_user | mwi_host     | orig_server_host |
>     orig_hostname |
>     +-----------------------+----------+--------------+----------------+-----------------------------------+-----------------+---------+------------+------------+-------------+---------------+--------------+----------+------------+--------------+--------------+--------------+----------+--------------+------------------+---------------+
>     | 83755-2574 at 10.10.55.1 <mailto:83755-2574 at 10.10.55.1> | 3755    
>     | 10.10.55.254 | 10.10.55.254   | "user"
>     <sip:3755 at 10.10.55.1:5060> <mailto:sip:3755 at 10.10.55.1:5060> |
>     Registered(UDP) | unknown | 1273141208 | SIPp/Linux | 3755       
>     | 172.30.30.129 | internal     | voip     | 10.10.55.1 |
>     5060         | 3755         | 10.10.55.254 | 3755     |
>     10.10.55.254 | 172.30.30.129    | voip          |
>     +-----------------------+----------+--------------+----------------+-----------------------------------+-----------------+---------+------------+------------+-------------+---------------+--------------+----------+------------+--------------+--------------+--------------+----------+--------------+------------------+---------------+
>
>     When I look in the sofia status profile, it seems that the correct
>     expires value is set. Any idea why the expires is not set
>     accordingly to the SIP REGISTER.
>
>     Moreover, when doing a "sofia status profile internal" with such a
>     huge subscribers list, the CPU is at 100% and processing no more
>     SIP messages.
>     Is there a way to prevent this, by setting for example a lower
>     priority to CLI request. I already mounted the SQLlite DB in RAM
>     to avoid IO bottlenecks.
>
>     regards,
>     rod
>
>
>     Le 04/05/2010 12:12, Rod. a ?crit :
>>     Hi,
>>
>>     I already thought about that, as I'm using static IP it could be
>>     even easier.
>>     But how to check network connectivity issue and reroute call to
>>     voicemail asap: call progress timeout ??
>>
>>     rod
>>
>>
>>
>>     Le 03/05/2010 21:42, Jan Berger a ?crit :
>>>     May a suggest a change filter developed if this really is needed?
>>>
>>>     Re-loading everything just in case something has changes is a
>>>     huge waste of bandwidth and CPU - if you install an intelligent
>>>     change filter you would be down to a few entries changing.
>>>
>>>     Jan
>>>
>>>     ------------------------------------------------------------------------
>>>     Date: Mon, 3 May 2010 21:06:54 +0400
>>>     From: kawarod at laposte.net <mailto:kawarod at laposte.net>
>>>     To: freeswitch-users at lists.freeswitch.org
>>>     <mailto:freeswitch-users at lists.freeswitch.org>
>>>     Subject: Re: [Freeswitch-users] Registration ODBC feeded by
>>>     another registrar proxy
>>>
>>>     Hi,
>>>
>>>     thanks for your answer and just some details to describe what
>>>     I'm looking for.
>>>     I have to register 25 000 subscribers, no NAT is involved, each
>>>     equipment has its own IP address.
>>>     These equipments are registering every 60 seconds on our current
>>>     platform, but I can change this parameter if needed.
>>>     Equipments are ADSL CPE (router), that's why I'm using 60sec
>>>     cause flapping could happen very often with ADSL if the copper
>>>     line is crappy. ADSL could be very unpredictable sometimes.
>>>     As I don't want to delay too much forwarding to voicemail if a
>>>     user is unavailable (network issue), 60 sec was chosen (bandwith
>>>     is not an issue). But as I told before, I'm open to your
>>>     suggestions.
>>>
>>>     To Philip, using a single SIP proxy (opensips/ser...) in front
>>>     of a FS cluster could be a single point of failure too.
>>>     I think that maybe a solution using DNS SRV to distribute the
>>>     load across a cluster could do the trick or some kind of LVS
>>>     (virtual IP shared across many servers)
>>>     XML curl is a good idea too.
>>>     To be honest, clustering is a must to avoid a single point of
>>>     failure, but FS performance as a SBC are really great even on
>>>     commodity hardware, more than 100 CallPerSecond with no
>>>     transcoding. That's why I think that a mix with a SIP registrar
>>>     and FS (and redundancy) could easily handle my 25 000
>>>     subscribers. I did some lab (one or 2 years ago) with Kamailio
>>>     registering 90 000 users every 60sec (1500 Registration per
>>>     second) without any issues.
>>>     In my network, 25 000 users are not pushing more than 10 CPS and
>>>     500 simultaneous call. I'm not doing VoIP termination.
>>>
>>>     At the moment, I'm just collecting data/feedback on what could
>>>     be done as I have some time to work on this project, and if
>>>     going further I will share the configuration as I did before:
>>>     http://wiki.freeswitch.org/wiki/SBC_Setup (not the best setup,
>>>     but hope it helps users to begin with FS)
>>>
>>>     regards,
>>>     rod.
>>>
>>>
>>>
>>>
>>>     Le 03/05/2010 19:54, David Ponzone a ?crit :
>>>
>>>         Rod,
>>>
>>>         Registering every 60 seconds is a bad idea, and this should
>>>         not be justified.
>>>         You should register every 1800 seconds and send a NAT
>>>         keepalive every X seconds.
>>>         X should be slightly lower than the NAT UDP timeout of the
>>>         router in front of the phones, if the phones are behind NAT.
>>>         If the phones are not behind NAT, NAT keepalive is not
>>>         necessary.
>>>
>>>         David Ponzone Direction Technique
>>>         email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>>>         tel:      01 74 03 18 97
>>>         gsm:   06 66 98 76 34
>>>
>>>         Service Client IPeva
>>>         tel:      0811 46 26 26
>>>         www.ipeva.fr  - www.ipeva-studio.com
>>>
>>>         /Ce message et toutes les pi?ces jointes sont confidentiels
>>>         et ?tablis ? l'intention exclusive de ses destinataires.
>>>         Toute utilisation ou diffusion non autoris?e est interdite.
>>>         Tout message ?lectronique est susceptible d'alt?ration.
>>>         /*/IPeva/*/ d?cline toute responsabilit? au titre de ce
>>>         message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous
>>>         n'?tes pas destinataire de ce message, merci de le d?truire
>>>         imm?diatement et d'avertir l'exp?diteur./
>>>         /
>>>         /
>>>
>>>
>>>
>>>         Le 03/05/2010 ? 15:39, Rod. a ?crit :
>>>
>>>             Hi list,
>>>
>>>             was playing with FS 1.0.6 and trying to test the
>>>             registration
>>>             performance of FS. (Yes I know FS is more suited as a
>>>             B2BUA, but please
>>>             read further :p)
>>>
>>>             So I did the following:
>>>                 - generate one xml file with 20 000 user account
>>>             like this:
>>>             <include>
>>>             <user id="1">
>>>             <params>
>>>             <param name="password" value="1234"/>
>>>             </params>
>>>             </user>
>>>             <user id="2">
>>>             <params>
>>>             <param name="password" value="1234"/>
>>>             </params>
>>>                         ...
>>>
>>>             Then I used Sipp to test how many registration per
>>>             second could be fired
>>>             to the server (quad core 2.83Ghz).
>>>             I setup ulimit variables, and disable nat.
>>>
>>>             I got this:
>>>                 - using SQL Lite: unable to get higher than 80
>>>             registrations per
>>>             second (in fact it's less than this number but didn't
>>>             test too much this
>>>             setup), I see a lot of retransmission in Sipp
>>>                 - using SQL Lite in ramdisk (tmpfs): OK with 80
>>>             registrations per
>>>             second but not much
>>>                 - using ODBC and mysql: 130 reg/sec is OK
>>>
>>>             With ODBC, above 150 reg/sec I see that FS is stalled to
>>>             100-110% CPU, I
>>>             think it's because I'm using only one SIP profile and
>>>             that SOFIA is
>>>             monothreaded for this SIP profile.
>>>             If I'd like to register every 60sec, the server has to
>>>             support at least
>>>             more than 300 registration per second.
>>>
>>>             So I'm wondering if I could setup something like this:
>>>                 - use another SIP Proxy as a registrar and feed the
>>>             ODBC
>>>             "sip_registration database" of FS
>>>                 - FS will be able to use this database to setup a call
>>>                 - use FS as the outbound proxy for call routing
>>>
>>>             But what about the user params that have been setup in
>>>             the xml file
>>>             above. I think that FS loads the user params each time a
>>>             user is registered.
>>>
>>>             Comments and advices are welcome.
>>>
>>>             regards,
>>>             rod.
>>>
>>>
>>>
>>>             _______________________________________________
>>>             FreeSWITCH-users mailing list
>>>             FreeSWITCH-users at lists.freeswitch.org
>>>             <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>             http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>             UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>             http://www.freeswitch.org <http://www.freeswitch.org/>
>>>
>>>
>>>
>>>         _______________________________________________
>>>         FreeSWITCH-users mailing list
>>>         FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>         http://www.freeswitch.org  <http://www.freeswitch.org/>
>>>            
>>>
>>>
>>>
>>>     ------------------------------------------------------------------------
>>>     Hotmail: Free, trusted and rich email service. Get it now.
>>>     <https://signup.live.com/signup.aspx?id=60969>
>>>
>>>
>>>     _______________________________________________
>>>     FreeSWITCH-users mailing list
>>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     http://www.freeswitch.org
>>>        
>>
>>
>>     _______________________________________________
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>>        
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    

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From stevendt at primrosebank.net  Fri May  7 04:00:47 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Fri, 7 May 2010 12:00:47 +0100
Subject: [Freeswitch-users] Sorting through a list of conditions -
	FreeSWITCH BooK ?
References: <017a01caec8a$ee74bb70$cb5e3250$@com><BE3F6D4F-9486-43F3-ACAF-70C84CB001FF@gmail.com><01b101caec94$1f009fd0$5d01df70$@com>
	<w2x87f2f3b91005051355j7c5c775dld43c79c6bf9d61cb@mail.gmail.com>
Message-ID: <A60A2CFD3B5C41D68FB970C1EB64DFF2@bp1.ad.bp.com>

Hi Mike,

was that comment a bit "tongue in cheek", or is there a FreeSWITCH book somewhere on the (near?) horizon ?

regards
Dave
  ----- Original Message ----- 
  From: Michael Collins 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, May 05, 2010 9:55 PM
  Subject: Re: [Freeswitch-users] Sorting through a list of conditions





  On Wed, May 5, 2010 at 1:47 PM, Ken Fulmer <kenfulmer at icstechnologysolutions.com> wrote:

    That is exactly what we needed. I?m still a little new to matching with regular expressions.

  You'll be interested in Chapter 5 of the upcoming FreeSWITCH book... :D
  -MC





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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From pjintheusa at gmail.com  Fri May  7 07:32:09 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Fri, 7 May 2010 10:32:09 -0400
Subject: [Freeswitch-users] recording problem
In-Reply-To: <n2k1fdf92721005062336g154dd71al2888b4253abfc4e2@mail.gmail.com>
References: <n2k1fdf92721005062336g154dd71al2888b4253abfc4e2@mail.gmail.com>
Message-ID: <t2q367751821005070732u91bd3669oeb792f6b2b169021@mail.gmail.com>

exec_after_bridge_app = Execute an application command after the bridge has
been terminated
http://wiki.freeswitch.org/wiki/Variable_exec_after_bridge_arg

Not sure when the correct call would be though....
execute_on_answer ??


On Fri, May 7, 2010 at 2:36 AM, babak yakhchali
<babak.freeswitch at gmail.com>wrote:

> Hi I'm using the code below to dispatch customers to operators in c#.
> everything is fine but recording is not taking place(I've tested both the
> hang up after bridge false and true).
> thanks
> public class Answering : IAppPlugin
>     {
>
>         ManagedSession Session;
>         public void Run(AppContext context)
>         {
>             Session = context.Session;
>             Session.HangupFunction = hangupHook;
>
>             if (. . .)      //no operator available wai
>             {
>                 Session.Answer();
>                 if (Session.Ready())
>                 {
>                     Session.StreamFile(m_sorryPrompt, -1);
>                     Session.SetVariable("sepanta_prompt_ended","true");
>
>                         Session.StreamFile(m_musicOnHold, -1);
>                 }
>             }//now u can talk
>             if (Session.Ready())
>             {
>                 string ticket = "1234";
>                 if (!Session.answered())
>                     Session.Answer();
>                 Session.Say(ticket, "en", "NUMBER", "PRONOUNCED",
> "FEMININE");
>                 Session.Execute("Playback",m_readyPrompt);
>                 Session.SetVariable("exec_after_bridge_app",
> "record_session");
>                 Session.SetVariable("exec_after_bridge_arg", recPath);
>
>  Session.Execute("bridge",string.Format("sofia/internal/{0}%{1}", op,
> Session.GetVariable("domain")));
>             }
>         }
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From william.suffill at gmail.com  Fri May  7 08:08:27 2010
From: william.suffill at gmail.com (William Suffill)
Date: Fri, 7 May 2010 11:08:27 -0400
Subject: [Freeswitch-users] Sorting through a list of conditions -
	FreeSWITCH BooK ?
In-Reply-To: <A60A2CFD3B5C41D68FB970C1EB64DFF2@bp1.ad.bp.com>
References: <017a01caec8a$ee74bb70$cb5e3250$@com>
	<BE3F6D4F-9486-43F3-ACAF-70C84CB001FF@gmail.com> 
	<01b101caec94$1f009fd0$5d01df70$@com>
	<w2x87f2f3b91005051355j7c5c775dld43c79c6bf9d61cb@mail.gmail.com>
	<A60A2CFD3B5C41D68FB970C1EB64DFF2@bp1.ad.bp.com>
Message-ID: <g2n6b65470d1005070808y30d1dce7q1c9b4fec8cc8f058@mail.gmail.com>

Yes. there is actually a book in the works. MC has more details. =)

-- W
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From douga at cachecomm.com  Fri May  7 10:29:06 2010
From: douga at cachecomm.com (Doug Albrechtsen)
Date: Fri, 07 May 2010 11:29:06 -0600
Subject: [Freeswitch-users] 2 B Channel Transfer (2BCT) and FreeSWITCH
In-Reply-To: <4BC4F7ED.50406@cachecomm.com>
References: <4BC4F7ED.50406@cachecomm.com>
Message-ID: <4BE44DE2.1050102@cachecomm.com>

Possibly a rephrase will help.  Can anyone help me with the procedure to 
initiate a 2B Channel Transfer on FreeSWITCH?

I see how it should be done in * with bridge...but can't seem to find 
much info on it for FreeSWITCH.  I see a function "pri_channel_bridge" 
in pri.c but I'm not sure where it is called form and how. 

I must be missing something super obvious...or not many people are using 
this feature.

Thanks.



Doug Albrechtsen wrote:
> Is anyone having success using 2B Channel Transfer with FreeSWITCH and 
> Sangoma?
>
> If so, what are the driver considerations and what does it take to make 
> the transfer work?
>
> Thanks,
> Doug
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From brian at freeswitch.org  Fri May  7 10:35:45 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 7 May 2010 12:35:45 -0500
Subject: [Freeswitch-users] 2 B Channel Transfer (2BCT) and FreeSWITCH
In-Reply-To: <4BE44DE2.1050102@cachecomm.com>
References: <4BC4F7ED.50406@cachecomm.com> <4BE44DE2.1050102@cachecomm.com>
Message-ID: <FD757212-87A3-4BDF-BAB0-1AE73D5F2FF1@freeswitch.org>

I don't know of any stack or endpoint using either libpri or sangoma's stack that can do a TBCT

/b

On May 7, 2010, at 12:29 PM, Doug Albrechtsen wrote:

> Possibly a rephrase will help.  Can anyone help me with the procedure to 
> initiate a 2B Channel Transfer on FreeSWITCH?
> 
> I see how it should be done in * with bridge...but can't seem to find 
> much info on it for FreeSWITCH.  I see a function "pri_channel_bridge" 
> in pri.c but I'm not sure where it is called form and how. 
> 
> I must be missing something super obvious...or not many people are using 
> this feature.
> 
> Thanks.
> 



From krzysztofdrewicz at gmail.com  Fri May  7 11:19:59 2010
From: krzysztofdrewicz at gmail.com (Krzysztof Drewicz)
Date: Fri, 7 May 2010 20:19:59 +0200
Subject: [Freeswitch-users] 2 B Channel Transfer (2BCT) and FreeSWITCH
In-Reply-To: <FD757212-87A3-4BDF-BAB0-1AE73D5F2FF1@freeswitch.org>
References: <4BC4F7ED.50406@cachecomm.com> <4BE44DE2.1050102@cachecomm.com>
	<FD757212-87A3-4BDF-BAB0-1AE73D5F2FF1@freeswitch.org>
Message-ID: <n2o9372a8331005071119lfd8f37f8v8c3140063492d22b@mail.gmail.com>

2010/5/7 Brian West <brian at freeswitch.org>:
> I don't know of any stack or endpoint using either libpri or sangoma's stack that can do a TBCT
>

Looks like * does do this job:

Status: Functional in DMS100 in Asterisk 1.4, and beta patch for 5ESS,
NI2 2BCT added into libpri trunk by mattf 2007-09-21
Contributors: mflorell (mflorell at-the-domain gmail.com)

http://bugs.digium.com/view.php?id=3554
http://www.misdn.org/index.php/Call_Forwarding
http://www.voip-info.org/wiki/view/Asterisk+cmd+BristuffZapCD
http://www.voip-info.org/wiki/view/Asterisk+cmd+SrxDeflect

; For digital ports using ISDN PRI protocols:
;   Support switch-side transfer (called 2BCT, RLT or other names)
;   This setting must be enabled on both ports involved, and the
;   'facilityenable' setting must also be enabled to allow sending
;   the transfer to the ISDN switch, since it sent in a FACILITY
;   message.
;
transfer=yes

But, this was coded in USA and for 5ESS, for europe i couldn't use it
with EuroISDN.
By the way, some guy (from Germany) did it for PRI connected between
enterprise PBX-grade Alcatel PBX (not Carrier grade) and asterisk like
2-3 years ago, only with chan_capi and AFAIR dialogic cards, as for
openzap/dahdi i havn't seen working solution by myself.

maybe commercial PRI stack from Sangoma is a good way to start?

also are you sure that 2BCT is required? maybe the 'path replacment'
or some QSIG variant singalling will be more appropriate?

>From my knowledge: some not-so-common ISDN features are realized by
different vendors in some VERY different ways (try to QSIG-GF Siemens
with Avaya or even Alcatel with Lucent...) so the question is 'is the
other side supporting this as well, and if it's compatible with my
isdn PRI stack'

wish you many luck on this, as i haven't found good and reliable
solution for this (vendor independent and 'other side'-independent).
I've been lurking for this functionality for years now.

Regards,


From brian at freeswitch.org  Fri May  7 11:27:10 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 7 May 2010 13:27:10 -0500
Subject: [Freeswitch-users] 2 B Channel Transfer (2BCT) and FreeSWITCH
In-Reply-To: <n2o9372a8331005071119lfd8f37f8v8c3140063492d22b@mail.gmail.com>
References: <4BC4F7ED.50406@cachecomm.com> <4BE44DE2.1050102@cachecomm.com>
	<FD757212-87A3-4BDF-BAB0-1AE73D5F2FF1@freeswitch.org>
	<n2o9372a8331005071119lfd8f37f8v8c3140063492d22b@mail.gmail.com>
Message-ID: <E56244FD-17F3-4560-8A1D-3BC335FC5D57@freeswitch.org>


On May 7, 2010, at 1:19 PM, Krzysztof Drewicz wrote:

> Looks like * does do this job:
> 
> Status: Functional in DMS100 in Asterisk 1.4, and beta patch for 5ESS,
> NI2 2BCT added into libpri trunk by mattf 2007-09-21
> Contributors: mflorell (mflorell at-the-domain gmail.com)
> 
> maybe commercial PRI stack from Sangoma is a good way to start?

Its on the list for Sangoma to add to to prid.

/b

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From douga at cachecomm.com  Fri May  7 11:51:40 2010
From: douga at cachecomm.com (Doug Albrechtsen)
Date: Fri, 07 May 2010 12:51:40 -0600
Subject: [Freeswitch-users] 2 B Channel Transfer (2BCT) and FreeSWITCH
In-Reply-To: <E56244FD-17F3-4560-8A1D-3BC335FC5D57@freeswitch.org>
References: <4BC4F7ED.50406@cachecomm.com>
	<4BE44DE2.1050102@cachecomm.com>	<FD757212-87A3-4BDF-BAB0-1AE73D5F2FF1@freeswitch.org>	<n2o9372a8331005071119lfd8f37f8v8c3140063492d22b@mail.gmail.com>
	<E56244FD-17F3-4560-8A1D-3BC335FC5D57@freeswitch.org>
Message-ID: <4BE4613C.4030805@cachecomm.com>

Yes...I believe it will be supported by them at some point.  However, 
I'm told by Sangoma that it is not on the short-term road map for the 
Sangoma PRI stack...but that if I want 2B Channel Transfer then I need 
libpri.  I'm still sorting through that, trying to figure out how to get 
it to go.

Thanks


Brian West wrote:
>
> On May 7, 2010, at 1:19 PM, Krzysztof Drewicz wrote:
>
>> Looks like * does do this job:
>>
>> Status: Functional in DMS100 in Asterisk 1.4, and beta patch for 5ESS,
>> NI2 2BCT added into libpri trunk by mattf 2007-09-21
>> Contributors: mflorell (mflorell at-the-domain gmail.com 
>> <http://gmail.com/>)
>>
>> maybe commercial PRI stack from Sangoma is a good way to start?
>
> Its on the list for Sangoma to add to to prid.
>
> /b
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From brian at freeswitch.org  Fri May  7 12:01:15 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 7 May 2010 14:01:15 -0500
Subject: [Freeswitch-users] 2 B Channel Transfer (2BCT) and FreeSWITCH
In-Reply-To: <4BE4613C.4030805@cachecomm.com>
References: <4BC4F7ED.50406@cachecomm.com>
	<4BE44DE2.1050102@cachecomm.com>	<FD757212-87A3-4BDF-BAB0-1AE73D5F2FF1@freeswitch.org>	<n2o9372a8331005071119lfd8f37f8v8c3140063492d22b@mail.gmail.com>
	<E56244FD-17F3-4560-8A1D-3BC335FC5D57@freeswitch.org>
	<4BE4613C.4030805@cachecomm.com>
Message-ID: <7699DCE3-4D79-48FF-BE44-52935D7634F0@freeswitch.org>

But work will have to be done in FreeSWITCH to send the indication to do the transfer its not magically going to happen.

/b

On May 7, 2010, at 1:51 PM, Doug Albrechtsen wrote:

> Yes...I believe it will be supported by them at some point.  However, 
> I'm told by Sangoma that it is not on the short-term road map for the 
> Sangoma PRI stack...but that if I want 2B Channel Transfer then I need 
> libpri.  I'm still sorting through that, trying to figure out how to get 
> it to go.
> 
> Thanks



From douga at cachecomm.com  Fri May  7 12:14:45 2010
From: douga at cachecomm.com (Doug Albrechtsen)
Date: Fri, 07 May 2010 13:14:45 -0600
Subject: [Freeswitch-users] 2 B Channel Transfer (2BCT) and FreeSWITCH
In-Reply-To: <7699DCE3-4D79-48FF-BE44-52935D7634F0@freeswitch.org>
References: <4BC4F7ED.50406@cachecomm.com>	<4BE44DE2.1050102@cachecomm.com>	<FD757212-87A3-4BDF-BAB0-1AE73D5F2FF1@freeswitch.org>	<n2o9372a8331005071119lfd8f37f8v8c3140063492d22b@mail.gmail.com>	<E56244FD-17F3-4560-8A1D-3BC335FC5D57@freeswitch.org>	<4BE4613C.4030805@cachecomm.com>
	<7699DCE3-4D79-48FF-BE44-52935D7634F0@freeswitch.org>
Message-ID: <4BE466A5.5040406@cachecomm.com>

I understand...Thanks for your reply.

Brian West wrote:
> But work will have to be done in FreeSWITCH to send the indication to do the transfer its not magically going to happen.
>
> /b
>
> On May 7, 2010, at 1:51 PM, Doug Albrechtsen wrote:
>
>   
>> Yes...I believe it will be supported by them at some point.  However, 
>> I'm told by Sangoma that it is not on the short-term road map for the 
>> Sangoma PRI stack...but that if I want 2B Channel Transfer then I need 
>> libpri.  I'm still sorting through that, trying to figure out how to get 
>> it to go.
>>
>> Thanks
>>     
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From msc at freeswitch.org  Fri May  7 14:55:25 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 7 May 2010 14:55:25 -0700
Subject: [Freeswitch-users] recording problem
In-Reply-To: <t2q367751821005070732u91bd3669oeb792f6b2b169021@mail.gmail.com>
References: <n2k1fdf92721005062336g154dd71al2888b4253abfc4e2@mail.gmail.com>
	<t2q367751821005070732u91bd3669oeb792f6b2b169021@mail.gmail.com>
Message-ID: <m2t87f2f3b91005071455jef076ee2g73b3990d52c91ffa@mail.gmail.com>

Look here:
http://wiki.freeswitch.org/wiki/Channel_Variables#Code_Execution_Related

There are some options for executing apps on either A or B leg just prior to
media being exchanged.

-MC

On Fri, May 7, 2010 at 7:32 AM, Phillip Jones <pjintheusa at gmail.com> wrote:

> exec_after_bridge_app = Execute an application command after the bridge
> has been terminated
> http://wiki.freeswitch.org/wiki/Variable_exec_after_bridge_arg
>
> Not sure when the correct call would be though....
> execute_on_answer ??
>
>
> On Fri, May 7, 2010 at 2:36 AM, babak yakhchali <
> babak.freeswitch at gmail.com> wrote:
>
>> Hi I'm using the code below to dispatch customers to operators in c#.
>> everything is fine but recording is not taking place(I've tested both the
>> hang up after bridge false and true).
>> thanks
>> public class Answering : IAppPlugin
>>     {
>>
>>         ManagedSession Session;
>>         public void Run(AppContext context)
>>         {
>>             Session = context.Session;
>>             Session.HangupFunction = hangupHook;
>>
>>             if (. . .)      //no operator available wai
>>             {
>>                 Session.Answer();
>>                 if (Session.Ready())
>>                 {
>>                     Session.StreamFile(m_sorryPrompt, -1);
>>                     Session.SetVariable("sepanta_prompt_ended","true");
>>
>>                         Session.StreamFile(m_musicOnHold, -1);
>>                 }
>>             }//now u can talk
>>             if (Session.Ready())
>>             {
>>                 string ticket = "1234";
>>                 if (!Session.answered())
>>                     Session.Answer();
>>                 Session.Say(ticket, "en", "NUMBER", "PRONOUNCED",
>> "FEMININE");
>>                 Session.Execute("Playback",m_readyPrompt);
>>                 Session.SetVariable("exec_after_bridge_app",
>> "record_session");
>>                 Session.SetVariable("exec_after_bridge_arg", recPath);
>>
>>  Session.Execute("bridge",string.Format("sofia/internal/{0}%{1}", op,
>> Session.GetVariable("domain")));
>>             }
>>         }
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Fri May  7 14:59:59 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 7 May 2010 14:59:59 -0700
Subject: [Freeswitch-users] Sorting through a list of conditions -
	FreeSWITCH BooK ?
In-Reply-To: <g2n6b65470d1005070808y30d1dce7q1c9b4fec8cc8f058@mail.gmail.com>
References: <017a01caec8a$ee74bb70$cb5e3250$@com>
	<BE3F6D4F-9486-43F3-ACAF-70C84CB001FF@gmail.com>
	<01b101caec94$1f009fd0$5d01df70$@com>
	<w2x87f2f3b91005051355j7c5c775dld43c79c6bf9d61cb@mail.gmail.com>
	<A60A2CFD3B5C41D68FB970C1EB64DFF2@bp1.ad.bp.com>
	<g2n6b65470d1005070808y30d1dce7q1c9b4fec8cc8f058@mail.gmail.com>
Message-ID: <i2g87f2f3b91005071459i16d8bd2fr7fa57a6304d2c0dc@mail.gmail.com>

It was, of course, tongue-in-cheek, but yes, there is a book forthcoming, if
some of my co-authors (cough Darren cough) would stop avoiding me. :)

http://www.packtpub.com/freeswitch-1-0-5-build-robust-high-performance-telephony-systems/book

Title is still in flux, but evidently the publisher is fond of flora cuz
he/she wants the agapanthus to stay on the cover. Stay tuned for a finalized
title/cover.

-MC

On Fri, May 7, 2010 at 8:08 AM, William Suffill
<william.suffill at gmail.com>wrote:

> Yes. there is actually a book in the works. MC has more details. =)
>
> -- W
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Fri May  7 15:04:30 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 7 May 2010 15:04:30 -0700
Subject: [Freeswitch-users] Fw: rfc2833 confusion
In-Reply-To: <247242E3584D4DBA852DFF721339D817@MOBILEE1705>
References: <B270322143E74F04A9EEF3359B31EB46@MOBILEE1705>
	<p2h367751821005061206w8eb9c3b1o5e5c25385048e822@mail.gmail.com>
	<247242E3584D4DBA852DFF721339D817@MOBILEE1705>
Message-ID: <h2j87f2f3b91005071504p9f4ae986rfd45bf373406531d@mail.gmail.com>

On Thu, May 6, 2010 at 12:23 PM, Madovsky <infos at madovsky.org> wrote:

>  right,
> but in case of the SIP caller needs to send digits to FS,
> his SIP phone should in any way send DTMF, isn't it ?
> in fact I'm trying to implement rfc2833 in my own sip phone through RTP,
> but if there is a way to send from caller a string var representing the
> digits typed to FS once bridge is done
> and transfer it as dtmf digits I will be happy too...
>
>

Hook up a phone to your FS. If you have the default dialplan then call 9999.
Go to fs_cli and make sure you are at /log level 7
>From the phone press some digits. You will see them pop up on the display:
2010-05-07 15:03:56.407327 [DEBUG] switch_rtp.c:2805 RTP RECV DTMF 1:1920
2010-05-07 15:03:56.407327 [DEBUG] mod_dptools.c:1409 Digit 1
2010-05-07 15:03:57.688237 [DEBUG] switch_rtp.c:2805 RTP RECV DTMF 2:2880
2010-05-07 15:03:57.688237 [DEBUG] mod_dptools.c:1409 Digit 2
2010-05-07 15:03:58.669178 [DEBUG] switch_rtp.c:2805 RTP RECV DTMF 5:3200
2010-05-07 15:03:58.669178 [DEBUG] mod_dptools.c:1409 Digit 5

FreeSWITCH knows to look for the DTMFs.
-MC

>  ----- Original Message -----
> *From:* Phillip Jones <pjintheusa at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Thursday, May 06, 2010 3:06 PM
> *Subject:* Re: [Freeswitch-users] Fw: rfc2833 confusion
>
> Yes that allows you to play a prompt and then wait for the person on the
> line to enter digits.
>
> On Thu, May 6, 2010 at 2:15 PM, Madovsky <infos at madovsky.org> wrote:
>
>>  is this the right link ?
>>
>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits
>>
>>
>>  ----- Original Message -----
>> *From:* Madovsky <infos at madovsky.org>
>> *To:* freeswitch-users at lists.freeswitch.org
>>   *Sent:* Thursday, May 06, 2010 2:10 PM
>> *Subject:* Re: [Freeswitch-users] rfc2833 confusion
>>
>> But how to catch digits form a dialplan when the call
>> is already on ?
>>
>> ----- Original Message -----
>> *From:* Phillip Jones <pjintheusa at gmail.com>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Sent:* Thursday, May 06, 2010 1:43 PM
>> *Subject:* Re: [Freeswitch-users] rfc2833 confusion
>>
>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_send_dtmf
>>
>> Don't get to hung up on the inband / out of band thing. This will be
>> negotiated by FS and the carrier when the call is set up. For instance, when
>> you use G729 you need out of band DTMF but for G711 you can use both. This
>> might change from call you call, depending on what media gateways can
>> handle.
>>
>> If you want to "see" and hear DTMF - fire up wireshark (or Tshark) and
>> capture the trace. You can see the RTP DTMF in the trace and hear the
>> clamped audio from the carrier perhaps. Or just hear the inband tones. It
>> gives a good visual of what is going on.
>>
>> On Thu, May 6, 2010 at 1:25 PM, Madovsky <infos at madovsky.org> wrote:
>>
>>>  I anyone knows a link of a programming example of how
>>> to inject rfc2833 in RTP flow (doesn't matter the language)
>>> It would be very useful
>>>
>>> Thanks
>>>
>>> F
>>>
>>>  ----- Original Message -----
>>> *From:* Madovsky <infos at madovsky.org>
>>> *To:* freeswitch-users at lists.freeswitch.org
>>>   *Sent:* Thursday, May 06, 2010 12:39 PM
>>> *Subject:* Re: [Freeswitch-users] rfc2833 confusion
>>>
>>> god, so many confusion for only an audio tone ! ;)
>>> anyway if I understand it's called inband by the fact that it's in RTP
>>> flow onlly.
>>> so the rfc2833 packet is only an event payload ...
>>>
>>>
>>> ----- Original Message -----
>>> *From:* Anthony Minessale <anthony.minessale at gmail.com>
>>> *To:* freeswitch-users at lists.freeswitch.org
>>> *Sent:* Thursday, May 06, 2010 11:52 AM
>>> *Subject:* Re: [Freeswitch-users] rfc2833 confusion
>>>
>>> both 2833 and actual tones in the audio stream are often called in-band
>>> because the 2833 is just a transport for the inband tones that is assumed to
>>> be reconstructed on the other side.  Some people refer to inband as the real
>>> tones and 2833 as not inband because it's not the tones.  So the one you
>>> have to look out for is people who call 2833 inband and do not clarify that
>>> it's inband packetized according the RFC2833
>>>
>>> The one that is not inband for sure is INFO (which has been dropped from
>>> the standard in favor of some ridiculous SUBSCRIBE based xml exchange)
>>>
>>> Like all things in SIP and anything else that comes with an RFC even if
>>> you know the answer you can't rely on trusting that answer because the real
>>> world will do whatever it wants and invalidate you. =D
>>>
>>>
>>> On Thu, May 6, 2010 at 10:36 AM, Madovsky <infos at madovsky.org> wrote:
>>>
>>>>  HI,
>>>>
>>>> is rfc2833 inband or outband ? is it only event or mixed event and sound
>>>> ?
>>>> I'm confused after read articles on this subject
>>>>
>>>> Thx
>>>>
>>>> F
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <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-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>  ------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From william.suffill at gmail.com  Fri May  7 16:02:51 2010
From: william.suffill at gmail.com (William Suffill)
Date: Fri, 7 May 2010 19:02:51 -0400
Subject: [Freeswitch-users] Sorting through a list of conditions -
	FreeSWITCH BooK ?
In-Reply-To: <i2g87f2f3b91005071459i16d8bd2fr7fa57a6304d2c0dc@mail.gmail.com>
References: <017a01caec8a$ee74bb70$cb5e3250$@com>
	<BE3F6D4F-9486-43F3-ACAF-70C84CB001FF@gmail.com> 
	<01b101caec94$1f009fd0$5d01df70$@com>
	<w2x87f2f3b91005051355j7c5c775dld43c79c6bf9d61cb@mail.gmail.com>
	<A60A2CFD3B5C41D68FB970C1EB64DFF2@bp1.ad.bp.com>
	<g2n6b65470d1005070808y30d1dce7q1c9b4fec8cc8f058@mail.gmail.com>
	<i2g87f2f3b91005071459i16d8bd2fr7fa57a6304d2c0dc@mail.gmail.com>
Message-ID: <k2o6b65470d1005071602te2cbd4d9z6f11b94dbc2445fd@mail.gmail.com>

What version of FreeSWITCH do we expect to be up to before release? LOL.
Look forward to seeing it tho.

-- W
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From infos at madovsky.org  Fri May  7 16:19:58 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 7 May 2010 19:19:58 -0400
Subject: [Freeswitch-users] Fw: rfc2833 confusion
References: <B270322143E74F04A9EEF3359B31EB46@MOBILEE1705><p2h367751821005061206w8eb9c3b1o5e5c25385048e822@mail.gmail.com><247242E3584D4DBA852DFF721339D817@MOBILEE1705>
	<h2j87f2f3b91005071504p9f4ae986rfd45bf373406531d@mail.gmail.com>
Message-ID: <A740F356C38942C7A5CD92EA5F3AD90B@MOBILEE1705>

I put on hold RFC2833 to try SIP INFO
but I got a 500 error from FS :

------------------------------------------------------------------------
   INFO sip:8519999999999 at 192.168.0.1:5080;transport=udp SIP/2.0
   Content-Length: 22
   Via: SIP/2.0/UDP 192.168.0.1:30082;rport;branch=z9hG4bKxEcVSs7CyRX5jWOWKwkAeQ..
   From: "FRANCK CHIONNA" <sip:8511234567890 at domain.com>;tag=93605822290
   To: <sip:8519999999999 at domaine.com>;tag=7ZQ8v6jvSpmZm
   Contact: <sip:8511234567890 at 192.168.0.1:30082>
   CSeq: 3 INFO
   Max-Forwards: 70
   Call-ID: 1279221367 at 192.168.0.1
   Content-Type: application/dtmf-relay

   Signal=1
   Duration=400
   ------------------------------------------------------------------------
tport_deliver(0x183cff0): msg 0x7f099c06fdf0 (476 bytes) from udp/192.168.0.1:5080/sip next=(nil)
nta: received INFO sip:8519999999999 at 192.168.0.1:5080;transport=udp SIP/2.0 (CSeq 3)
nta: INFO (3) going to existing INFO transaction
nta: timer I fired, terminate 407 response
nta_incoming_timer: 0/0 resent, 0/0 tout, 1/2 term, 1/3 free
nta: timer I fired, terminate 200 response
nta_incoming_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/2 free
tport_wakeup_pri(0x183cff0): events IN
tport_recv_event(0x183cff0)
tport_recv_iovec(0x183cff0) msg 0x7f099c06fdf0 from (udp/192.168.0.1:5080) has 476 bytes, veclen = 1
recv 476 bytes from udp/[192.168.0.1]:30082 at 23:15:15.203315:

is the SIP message malformatted ?
can't see why it fails

Thanks

Franck

  ----- Original Message ----- 
  From: Michael Collins 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, May 07, 2010 6:04 PM
  Subject: Re: [Freeswitch-users] Fw: rfc2833 confusion





  On Thu, May 6, 2010 at 12:23 PM, Madovsky <infos at madovsky.org> wrote:

    right,
    but in case of the SIP caller needs to send digits to FS,
    his SIP phone should in any way send DTMF, isn't it ?
    in fact I'm trying to implement rfc2833 in my own sip phone through RTP,
    but if there is a way to send from caller a string var representing the digits typed to FS once bridge is done
    and transfer it as dtmf digits I will be happy too... 


  Hook up a phone to your FS. If you have the default dialplan then call 9999. Go to fs_cli and make sure you are at /log level 7  
  From the phone press some digits. You will see them pop up on the display:
  2010-05-07 15:03:56.407327 [DEBUG] switch_rtp.c:2805 RTP RECV DTMF 1:1920
  2010-05-07 15:03:56.407327 [DEBUG] mod_dptools.c:1409 Digit 1
  2010-05-07 15:03:57.688237 [DEBUG] switch_rtp.c:2805 RTP RECV DTMF 2:2880
  2010-05-07 15:03:57.688237 [DEBUG] mod_dptools.c:1409 Digit 2
  2010-05-07 15:03:58.669178 [DEBUG] switch_rtp.c:2805 RTP RECV DTMF 5:3200
  2010-05-07 15:03:58.669178 [DEBUG] mod_dptools.c:1409 Digit 5

  FreeSWITCH knows to look for the DTMFs.
  -MC

      ----- Original Message ----- 
      From: Phillip Jones 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Thursday, May 06, 2010 3:06 PM
      Subject: Re: [Freeswitch-users] Fw: rfc2833 confusion


      Yes that allows you to play a prompt and then wait for the person on the line to enter digits.


      On Thu, May 6, 2010 at 2:15 PM, Madovsky <infos at madovsky.org> wrote:

        is this the right link ?

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


        ----- Original Message ----- 
        From: Madovsky 
        To: freeswitch-users at lists.freeswitch.org 
        Sent: Thursday, May 06, 2010 2:10 PM
        Subject: Re: [Freeswitch-users] rfc2833 confusion


        But how to catch digits form a dialplan when the call
        is already on ?
          ----- Original Message ----- 
          From: Phillip Jones 
          To: freeswitch-users at lists.freeswitch.org 
          Sent: Thursday, May 06, 2010 1:43 PM
          Subject: Re: [Freeswitch-users] rfc2833 confusion


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

          Don't get to hung up on the inband / out of band thing. This will be negotiated by FS and the carrier when the call is set up. For instance, when you use G729 you need out of band DTMF but for G711 you can use both. This might change from call you call, depending on what media gateways can handle.

          If you want to "see" and hear DTMF - fire up wireshark (or Tshark) and capture the trace. You can see the RTP DTMF in the trace and hear the clamped audio from the carrier perhaps. Or just hear the inband tones. It gives a good visual of what is going on.


          On Thu, May 6, 2010 at 1:25 PM, Madovsky <infos at madovsky.org> wrote:

            I anyone knows a link of a programming example of how
            to inject rfc2833 in RTP flow (doesn't matter the language)
            It would be very useful

            Thanks

            F
              ----- Original Message ----- 
              From: Madovsky 
              To: freeswitch-users at lists.freeswitch.org 
              Sent: Thursday, May 06, 2010 12:39 PM
              Subject: Re: [Freeswitch-users] rfc2833 confusion


              god, so many confusion for only an audio tone ! ;)
              anyway if I understand it's called inband by the fact that it's in RTP flow onlly.
              so the rfc2833 packet is only an event payload ...

                ----- Original Message ----- 
                From: Anthony Minessale 
                To: freeswitch-users at lists.freeswitch.org 
                Sent: Thursday, May 06, 2010 11:52 AM
                Subject: Re: [Freeswitch-users] rfc2833 confusion


                both 2833 and actual tones in the audio stream are often called in-band because the 2833 is just a transport for the inband tones that is assumed to be reconstructed on the other side.  Some people refer to inband as the real tones and 2833 as not inband because it's not the tones.  So the one you have to look out for is people who call 2833 inband and do not clarify that it's inband packetized according the RFC2833  


                The one that is not inband for sure is INFO (which has been dropped from the standard in favor of some ridiculous SUBSCRIBE based xml exchange) 


                Like all things in SIP and anything else that comes with an RFC even if you know the answer you can't rely on trusting that answer because the real world will do whatever it wants and invalidate you. =D




                On Thu, May 6, 2010 at 10:36 AM, Madovsky <infos at madovsky.org> wrote:

                  HI,

                  is rfc2833 inband or outband ? is it only event or mixed event and sound ?
                  I'm confused after read articles on this subject

                  Thx

                  F

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





                -- 
                Anthony Minessale II

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

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

                FreeSWITCH Developer Conference
                sip:888 at conference.freeswitch.org
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From infos at madovsky.org  Fri May  7 16:38:14 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 7 May 2010 19:38:14 -0400
Subject: [Freeswitch-users] Fw: rfc2833 confusion
References: <B270322143E74F04A9EEF3359B31EB46@MOBILEE1705><p2h367751821005061206w8eb9c3b1o5e5c25385048e822@mail.gmail.com><247242E3584D4DBA852DFF721339D817@MOBILEE1705>
	<h2j87f2f3b91005071504p9f4ae986rfd45bf373406531d@mail.gmail.com>
Message-ID: <7BEF0B9210FC496FBEE72A87AA1B349A@MOBILEE1705>

Mike 
about RFC2833 I set internal.xml in debug mode
with log level 7 but can't see any line like below with
a working phone

Thx
  ----- Original Message ----- 
  From: Michael Collins 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, May 07, 2010 6:04 PM
  Subject: Re: [Freeswitch-users] Fw: rfc2833 confusion





  On Thu, May 6, 2010 at 12:23 PM, Madovsky <infos at madovsky.org> wrote:

    right,
    but in case of the SIP caller needs to send digits to FS,
    his SIP phone should in any way send DTMF, isn't it ?
    in fact I'm trying to implement rfc2833 in my own sip phone through RTP,
    but if there is a way to send from caller a string var representing the digits typed to FS once bridge is done
    and transfer it as dtmf digits I will be happy too... 


  Hook up a phone to your FS. If you have the default dialplan then call 9999. Go to fs_cli and make sure you are at /log level 7  
  From the phone press some digits. You will see them pop up on the display:
  2010-05-07 15:03:56.407327 [DEBUG] switch_rtp.c:2805 RTP RECV DTMF 1:1920
  2010-05-07 15:03:56.407327 [DEBUG] mod_dptools.c:1409 Digit 1
  2010-05-07 15:03:57.688237 [DEBUG] switch_rtp.c:2805 RTP RECV DTMF 2:2880
  2010-05-07 15:03:57.688237 [DEBUG] mod_dptools.c:1409 Digit 2
  2010-05-07 15:03:58.669178 [DEBUG] switch_rtp.c:2805 RTP RECV DTMF 5:3200
  2010-05-07 15:03:58.669178 [DEBUG] mod_dptools.c:1409 Digit 5

  FreeSWITCH knows to look for the DTMFs.
  -MC

      ----- Original Message ----- 
      From: Phillip Jones 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Thursday, May 06, 2010 3:06 PM
      Subject: Re: [Freeswitch-users] Fw: rfc2833 confusion


      Yes that allows you to play a prompt and then wait for the person on the line to enter digits.


      On Thu, May 6, 2010 at 2:15 PM, Madovsky <infos at madovsky.org> wrote:

        is this the right link ?

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


        ----- Original Message ----- 
        From: Madovsky 
        To: freeswitch-users at lists.freeswitch.org 
        Sent: Thursday, May 06, 2010 2:10 PM
        Subject: Re: [Freeswitch-users] rfc2833 confusion


        But how to catch digits form a dialplan when the call
        is already on ?
          ----- Original Message ----- 
          From: Phillip Jones 
          To: freeswitch-users at lists.freeswitch.org 
          Sent: Thursday, May 06, 2010 1:43 PM
          Subject: Re: [Freeswitch-users] rfc2833 confusion


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

          Don't get to hung up on the inband / out of band thing. This will be negotiated by FS and the carrier when the call is set up. For instance, when you use G729 you need out of band DTMF but for G711 you can use both. This might change from call you call, depending on what media gateways can handle.

          If you want to "see" and hear DTMF - fire up wireshark (or Tshark) and capture the trace. You can see the RTP DTMF in the trace and hear the clamped audio from the carrier perhaps. Or just hear the inband tones. It gives a good visual of what is going on.


          On Thu, May 6, 2010 at 1:25 PM, Madovsky <infos at madovsky.org> wrote:

            I anyone knows a link of a programming example of how
            to inject rfc2833 in RTP flow (doesn't matter the language)
            It would be very useful

            Thanks

            F
              ----- Original Message ----- 
              From: Madovsky 
              To: freeswitch-users at lists.freeswitch.org 
              Sent: Thursday, May 06, 2010 12:39 PM
              Subject: Re: [Freeswitch-users] rfc2833 confusion


              god, so many confusion for only an audio tone ! ;)
              anyway if I understand it's called inband by the fact that it's in RTP flow onlly.
              so the rfc2833 packet is only an event payload ...

                ----- Original Message ----- 
                From: Anthony Minessale 
                To: freeswitch-users at lists.freeswitch.org 
                Sent: Thursday, May 06, 2010 11:52 AM
                Subject: Re: [Freeswitch-users] rfc2833 confusion


                both 2833 and actual tones in the audio stream are often called in-band because the 2833 is just a transport for the inband tones that is assumed to be reconstructed on the other side.  Some people refer to inband as the real tones and 2833 as not inband because it's not the tones.  So the one you have to look out for is people who call 2833 inband and do not clarify that it's inband packetized according the RFC2833  


                The one that is not inband for sure is INFO (which has been dropped from the standard in favor of some ridiculous SUBSCRIBE based xml exchange) 


                Like all things in SIP and anything else that comes with an RFC even if you know the answer you can't rely on trusting that answer because the real world will do whatever it wants and invalidate you. =D




                On Thu, May 6, 2010 at 10:36 AM, Madovsky <infos at madovsky.org> wrote:

                  HI,

                  is rfc2833 inband or outband ? is it only event or mixed event and sound ?
                  I'm confused after read articles on this subject

                  Thx

                  F

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                -- 
                Anthony Minessale II

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

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

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



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From msc at freeswitch.org  Fri May  7 16:50:09 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 7 May 2010 16:50:09 -0700
Subject: [Freeswitch-users] Fw: rfc2833 confusion
In-Reply-To: <7BEF0B9210FC496FBEE72A87AA1B349A@MOBILEE1705>
References: <B270322143E74F04A9EEF3359B31EB46@MOBILEE1705>
	<p2h367751821005061206w8eb9c3b1o5e5c25385048e822@mail.gmail.com>
	<247242E3584D4DBA852DFF721339D817@MOBILEE1705>
	<h2j87f2f3b91005071504p9f4ae986rfd45bf373406531d@mail.gmail.com>
	<7BEF0B9210FC496FBEE72A87AA1B349A@MOBILEE1705>
Message-ID: <h2m87f2f3b91005071650qa9c7e67cwe3553b512857f590@mail.gmail.com>

On Fri, May 7, 2010 at 4:38 PM, Madovsky <infos at madovsky.org> wrote:

>  Mike
> about RFC2833 I set internal.xml in debug mode
> with log level 7 but can't see any line like below with
> a working phone
>

Then it's not really a "working" phone or something is wrong with your
setup. I would backup your configs, then re-run "make samples" and start
from scratch. With a single phone and nothing else connected to FS, try
again. If it still does not work then you've got an issue outside of
FreeSWITCH and you will need to break out the tools like tcpdump and
wireshark to see what's going on.

-MC
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From infos at madovsky.org  Fri May  7 17:08:02 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 7 May 2010 20:08:02 -0400
Subject: [Freeswitch-users] Fw: rfc2833 confusion
References: <B270322143E74F04A9EEF3359B31EB46@MOBILEE1705><p2h367751821005061206w8eb9c3b1o5e5c25385048e822@mail.gmail.com><247242E3584D4DBA852DFF721339D817@MOBILEE1705><h2j87f2f3b91005071504p9f4ae986rfd45bf373406531d@mail.gmail.com><7BEF0B9210FC496FBEE72A87AA1B349A@MOBILEE1705>
	<h2m87f2f3b91005071650qa9c7e67cwe3553b512857f590@mail.gmail.com>
Message-ID: <6A393898BF4D465BB35702727D517DCD@MOBILEE1705>

I did, the problem is my code only.
with other sip phone like x-lite it works...
with tcpdump and wireshark I can't see any strange packets...
anyway I think I will use another emergency solution for now.

Thx

F
  ----- Original Message ----- 
  From: Michael Collins 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, May 07, 2010 7:50 PM
  Subject: Re: [Freeswitch-users] Fw: rfc2833 confusion





  On Fri, May 7, 2010 at 4:38 PM, Madovsky <infos at madovsky.org> wrote:

    Mike 
    about RFC2833 I set internal.xml in debug mode
    with log level 7 but can't see any line like below with
    a working phone

  Then it's not really a "working" phone or something is wrong with your setup. I would backup your configs, then re-run "make samples" and start from scratch. With a single phone and nothing else connected to FS, try again. If it still does not work then you've got an issue outside of FreeSWITCH and you will need to break out the tools like tcpdump and wireshark to see what's going on. 

  -MC





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From infos at madovsky.org  Fri May  7 17:32:11 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 7 May 2010 20:32:11 -0400
Subject: [Freeswitch-users] leg B uuid
Message-ID: <081C33E659024A07A7FAFFE18EAF3CBB@MOBILEE1705>

Is there an already exsting variable which represent
the UUID of B leg from the A leg phone once the call is established ?
(sorry Mathieu I lost all my notes of this subject yesterday)

Thanks
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From mrene_lists at avgs.ca  Fri May  7 17:34:13 2010
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Fri, 7 May 2010 20:34:13 -0400
Subject: [Freeswitch-users] leg B uuid
In-Reply-To: <081C33E659024A07A7FAFFE18EAF3CBB@MOBILEE1705>
References: <081C33E659024A07A7FAFFE18EAF3CBB@MOBILEE1705>
Message-ID: <B7667045-D5EF-4885-8548-BF7788053E3B@avgs.ca>

signal_bond

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




On 2010-05-07, at 8:32 PM, Madovsky wrote:

> Is there an already exsting variable which represent
> the UUID of B leg from the A leg phone once the call is established ?
> (sorry Mathieu I lost all my notes of this subject yesterday)
>  
> Thanks
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From infos at madovsky.org  Fri May  7 17:43:54 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 7 May 2010 20:43:54 -0400
Subject: [Freeswitch-users] leg B uuid
References: <081C33E659024A07A7FAFFE18EAF3CBB@MOBILEE1705>
	<B7667045-D5EF-4885-8548-BF7788053E3B@avgs.ca>
Message-ID: <78103F23DF3240609228BBC9933DAD27@MOBILEE1705>

ok thanks,
if it's a channel variable so it's not on wiki
  ----- Original Message ----- 
  From: Mathieu Rene 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, May 07, 2010 8:34 PM
  Subject: Re: [Freeswitch-users] leg B uuid


  signal_bond


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








  On 2010-05-07, at 8:32 PM, Madovsky wrote:


    Is there an already exsting variable which represent
    the UUID of B leg from the A leg phone once the call is established ?
    (sorry Mathieu I lost all my notes of this subject yesterday)

    Thanks
    _______________________________________________
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    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From dujinfang at gmail.com  Fri May  7 18:42:04 2010
From: dujinfang at gmail.com (Seven Du)
Date: Sat, 8 May 2010 09:42:04 +0800
Subject: [Freeswitch-users] Sorting through a list of conditions -
	FreeSWITCH BooK ?
In-Reply-To: <i2g87f2f3b91005071459i16d8bd2fr7fa57a6304d2c0dc@mail.gmail.com>
References: <017a01caec8a$ee74bb70$cb5e3250$@com>
	<BE3F6D4F-9486-43F3-ACAF-70C84CB001FF@gmail.com>
	<01b101caec94$1f009fd0$5d01df70$@com>
	<w2x87f2f3b91005051355j7c5c775dld43c79c6bf9d61cb@mail.gmail.com>
	<A60A2CFD3B5C41D68FB970C1EB64DFF2@bp1.ad.bp.com>
	<g2n6b65470d1005070808y30d1dce7q1c9b4fec8cc8f058@mail.gmail.com>
	<i2g87f2f3b91005071459i16d8bd2fr7fa57a6304d2c0dc@mail.gmail.com>
Message-ID: <h2k23f91031005071842t865d2765x595a7a2572f13638@mail.gmail.com>

Great to see this. In the mean time, I also started to "write" a
FreeSWITCH book a short time ago. Obviously it's a Chinese version
however those who interested in that can get a glance by the help of
google translate. I had thought to announce that publicly when I have
more chapters. I haven't though to publish it because there are not
many FreeSWITCH users in China,  currently it's available in a
CC-BY-NC-ND license.

The English name of the book is FreeSWITCH: VoIP in Action. It's not a
good name but I want both FreeSWITCH and VoIP in it. I started in
Chapter two which was inspired(I borrowed some, but more than 70% was
re-written, I tried to publish it on an offline magazine but that
didn't happen ) by an article M.C. wrote a few months ago.

http://www.freeswitch.org.cn/document

Obviously the book will be much less professional than the official
one which I also eager to get a copy. I'd like to translate it into
Chinese if it is published. There are not many readers in China
currently but I believe it will has a better future - both FreeSWITCH
and the book.


2010/5/8 Michael Collins <msc at freeswitch.org>:
> It was, of course, tongue-in-cheek, but yes, there is a book forthcoming, if
> some of my co-authors (cough Darren cough) would stop avoiding me. :)
>
> http://www.packtpub.com/freeswitch-1-0-5-build-robust-high-performance-telephony-systems/book
>
> Title is still in flux, but evidently the publisher is fond of flora cuz
> he/she wants the agapanthus to stay on the cover. Stay tuned for a finalized
> title/cover.
>
> -MC
>
> On Fri, May 7, 2010 at 8:08 AM, William Suffill <william.suffill at gmail.com>
> wrote:
>>
>> Yes. there is actually a book in the works. MC has more details. =)
>> -- W
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



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


From bwibowo at gmail.com  Fri May  7 19:04:30 2010
From: bwibowo at gmail.com (budi wibowo)
Date: Sat, 8 May 2010 09:04:30 +0700
Subject: [Freeswitch-users] outgoing gateway
In-Reply-To: <472BC89E-11F0-49C2-8EA5-23783BC38770@jerris.com>
References: <l2o547ec34a1005061642y37cd9880tac6e3aef1da1be1@mail.gmail.com>
	<t2u94f7dfb11005062034wc952a56ev92417d4456cc8b55@mail.gmail.com>
	<1077559925-1273204385-cardhu_decombobulator_blackberry.rim.net-1446324892-@bda057.bisx.prodap.on.blackberry>
	<472BC89E-11F0-49C2-8EA5-23783BC38770@jerris.com>
Message-ID: <z2y547ec34a1005071904mcc2aa5d6wbec359c34d639f61@mail.gmail.com>

thx, after reading the document i still got confusion about sofia module
i give example below
Dialing Through A Gateway(SIP Provider)

A gateway is a means for making outbound calls through a SIP provider. For
example:
sofia/gateway/mygateway.com/1234


by default i dont find any directory named sofia, where i should put this
directory?
many document i read telling about sofia

TIA

budi


On Fri, May 7, 2010 at 12:55 PM, Michael Jerris <mike at jerris.com> wrote:

> http://wiki.freeswitch.org/wiki/Dialplan
> <http://wiki.freeswitch.org/wiki/Dialplan>
> On May 6, 2010, at 11:53 PM, Budi wibowo wrote:
>
> Yes I have siphit installed, I tried to make some changes on dialplan file
> but call always goes to other server, what should I configure to implement
> this outgoing call
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dujinfang at gmail.com  Fri May  7 19:26:48 2010
From: dujinfang at gmail.com (Seven Du)
Date: Sat, 8 May 2010 10:26:48 +0800
Subject: [Freeswitch-users] outgoing gateway
In-Reply-To: <z2y547ec34a1005071904mcc2aa5d6wbec359c34d639f61@mail.gmail.com>
References: <l2o547ec34a1005061642y37cd9880tac6e3aef1da1be1@mail.gmail.com>
	<t2u94f7dfb11005062034wc952a56ev92417d4456cc8b55@mail.gmail.com>
	<1077559925-1273204385-cardhu_decombobulator_blackberry.rim.net-1446324892-@bda057.bisx.prodap.on.blackberry>
	<472BC89E-11F0-49C2-8EA5-23783BC38770@jerris.com>
	<z2y547ec34a1005071904mcc2aa5d6wbec359c34d639f61@mail.gmail.com>
Message-ID: <t2t23f91031005071926l451137ffo76d667030f41f573@mail.gmail.com>

default configuration files of mod_sofia is in conf/sip_profiles

http://wiki.freeswitch.org/wiki/Sofia
http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files

2010/5/8 budi wibowo <bwibowo at gmail.com>:
> thx, after reading the document i still got confusion about sofia module
> i give example below
>
> Dialing Through A Gateway(SIP Provider)
>
> A gateway is a means for making outbound calls through a SIP provider. For
> example:
>
> sofia/gateway/mygateway.com/1234
>
> by default i dont find any directory named sofia, where i should put this
> directory?
> many document i read telling about sofia
> TIA
> budi
>
> On Fri, May 7, 2010 at 12:55 PM, Michael Jerris <mike at jerris.com> wrote:
>>
>> http://wiki.freeswitch.org/wiki/Dialplan
>> On May 6, 2010, at 11:53 PM, Budi wibowo wrote:
>>
>> Yes I have siphit installed, I tried to make some changes on dialplan file
>> but call always goes to other server, what should I configure to implement
>> this outgoing call
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



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


From dujinfang at gmail.com  Fri May  7 19:30:29 2010
From: dujinfang at gmail.com (Seven Du)
Date: Sat, 8 May 2010 10:30:29 +0800
Subject: [Freeswitch-users] outgoing gateway
In-Reply-To: <t2t23f91031005071926l451137ffo76d667030f41f573@mail.gmail.com>
References: <l2o547ec34a1005061642y37cd9880tac6e3aef1da1be1@mail.gmail.com>
	<t2u94f7dfb11005062034wc952a56ev92417d4456cc8b55@mail.gmail.com>
	<1077559925-1273204385-cardhu_decombobulator_blackberry.rim.net-1446324892-@bda057.bisx.prodap.on.blackberry>
	<472BC89E-11F0-49C2-8EA5-23783BC38770@jerris.com>
	<z2y547ec34a1005071904mcc2aa5d6wbec359c34d639f61@mail.gmail.com>
	<t2t23f91031005071926l451137ffo76d667030f41f573@mail.gmail.com>
Message-ID: <y2z23f91031005071930p3f96278aj9f7db3a5e0df60aa@mail.gmail.com>

In the mean time, I think it's a good article for you:

http://www.linuxpromagazine.com/Issues/2009/106/TALK-SOFT

2010/5/8 Seven Du <dujinfang at gmail.com>:
> default configuration files of mod_sofia is in conf/sip_profiles
>
> http://wiki.freeswitch.org/wiki/Sofia
> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files
>
> 2010/5/8 budi wibowo <bwibowo at gmail.com>:
>> thx, after reading the document i still got confusion about sofia module
>> i give example below
>>
>> Dialing Through A Gateway(SIP Provider)
>>
>> A gateway is a means for making outbound calls through a SIP provider. For
>> example:
>>
>> sofia/gateway/mygateway.com/1234
>>
>> by default i dont find any directory named sofia, where i should put this
>> directory?
>> many document i read telling about sofia
>> TIA
>> budi
>>
>> On Fri, May 7, 2010 at 12:55 PM, Michael Jerris <mike at jerris.com> wrote:
>>>
>>> http://wiki.freeswitch.org/wiki/Dialplan
>>> On May 6, 2010, at 11:53 PM, Budi wibowo wrote:
>>>
>>> Yes I have siphit installed, I tried to make some changes on dialplan file
>>> but call always goes to other server, what should I configure to implement
>>> this outgoing call
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Blog: http://www.dujinfang.com
> Proj: ?http://www.freeswitch.org.cn
>



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


From babak.freeswitch at gmail.com  Fri May  7 21:12:07 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Sat, 8 May 2010 08:42:07 +0430
Subject: [Freeswitch-users] recording problem
In-Reply-To: <m2t87f2f3b91005071455jef076ee2g73b3990d52c91ffa@mail.gmail.com>
References: <n2k1fdf92721005062336g154dd71al2888b4253abfc4e2@mail.gmail.com>
	<t2q367751821005070732u91bd3669oeb792f6b2b169021@mail.gmail.com>
	<m2t87f2f3b91005071455jef076ee2g73b3990d52c91ffa@mail.gmail.com>
Message-ID: <AANLkTilb8SmouZOVajLBD35a1i0Hh9BFVIrazKOKtcIw@mail.gmail.com>

I've tested execute on answer but it's not working when the call is answered
I read that I can use nolocal and export but it is not working too
thanks
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From babak.freeswitch at gmail.com  Fri May  7 21:37:20 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Sat, 8 May 2010 09:07:20 +0430
Subject: [Freeswitch-users] recording problem
In-Reply-To: <AANLkTilb8SmouZOVajLBD35a1i0Hh9BFVIrazKOKtcIw@mail.gmail.com>
References: <n2k1fdf92721005062336g154dd71al2888b4253abfc4e2@mail.gmail.com>
	<t2q367751821005070732u91bd3669oeb792f6b2b169021@mail.gmail.com>
	<m2t87f2f3b91005071455jef076ee2g73b3990d52c91ffa@mail.gmail.com>
	<AANLkTilb8SmouZOVajLBD35a1i0Hh9BFVIrazKOKtcIw@mail.gmail.com>
Message-ID: <AANLkTikCVLf-_YKOPQr6ofnIITi1WmS0sCUMOD8QI5wS@mail.gmail.com>

Hi
I've used the bridge_pre_execute_bleg_app and it is working for me
thanks
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From msc at freeswitch.org  Fri May  7 22:46:16 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 7 May 2010 22:46:16 -0700
Subject: [Freeswitch-users] Sorting through a list of conditions -
	FreeSWITCH BooK ?
In-Reply-To: <h2k23f91031005071842t865d2765x595a7a2572f13638@mail.gmail.com>
References: <017a01caec8a$ee74bb70$cb5e3250$@com>
	<BE3F6D4F-9486-43F3-ACAF-70C84CB001FF@gmail.com>
	<01b101caec94$1f009fd0$5d01df70$@com>
	<w2x87f2f3b91005051355j7c5c775dld43c79c6bf9d61cb@mail.gmail.com>
	<A60A2CFD3B5C41D68FB970C1EB64DFF2@bp1.ad.bp.com>
	<g2n6b65470d1005070808y30d1dce7q1c9b4fec8cc8f058@mail.gmail.com>
	<i2g87f2f3b91005071459i16d8bd2fr7fa57a6304d2c0dc@mail.gmail.com>
	<h2k23f91031005071842t865d2765x595a7a2572f13638@mail.gmail.com>
Message-ID: <z2o87f2f3b91005072246y39206088g6900a8a81434626c@mail.gmail.com>

Thanks Seven!
-MC

On Fri, May 7, 2010 at 6:42 PM, Seven Du <dujinfang at gmail.com> wrote:

> Great to see this. In the mean time, I also started to "write" a
> FreeSWITCH book a short time ago. Obviously it's a Chinese version
> however those who interested in that can get a glance by the help of
> google translate. I had thought to announce that publicly when I have
> more chapters. I haven't though to publish it because there are not
> many FreeSWITCH users in China,  currently it's available in a
> CC-BY-NC-ND license.
>
> The English name of the book is FreeSWITCH: VoIP in Action. It's not a
> good name but I want both FreeSWITCH and VoIP in it. I started in
> Chapter two which was inspired(I borrowed some, but more than 70% was
> re-written, I tried to publish it on an offline magazine but that
> didn't happen ) by an article M.C. wrote a few months ago.
>
> http://www.freeswitch.org.cn/document
>
> Obviously the book will be much less professional than the official
> one which I also eager to get a copy. I'd like to translate it into
> Chinese if it is published. There are not many readers in China
> currently but I believe it will has a better future - both FreeSWITCH
> and the book.
>
>
> 2010/5/8 Michael Collins <msc at freeswitch.org>:
> > It was, of course, tongue-in-cheek, but yes, there is a book forthcoming,
> if
> > some of my co-authors (cough Darren cough) would stop avoiding me. :)
> >
> >
> http://www.packtpub.com/freeswitch-1-0-5-build-robust-high-performance-telephony-systems/book
> >
> > Title is still in flux, but evidently the publisher is fond of flora cuz
> > he/she wants the agapanthus to stay on the cover. Stay tuned for a
> finalized
> > title/cover.
> >
> > -MC
> >
> > On Fri, May 7, 2010 at 8:08 AM, William Suffill <
> william.suffill at gmail.com>
> > wrote:
> >>
> >> Yes. there is actually a book in the works. MC has more details. =)
> >> -- W
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Blog: http://www.dujinfang.com
> Proj:  http://www.freeswitch.org.cn
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Fri May  7 22:50:29 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 7 May 2010 22:50:29 -0700
Subject: [Freeswitch-users] leg B uuid
In-Reply-To: <B7667045-D5EF-4885-8548-BF7788053E3B@avgs.ca>
References: <081C33E659024A07A7FAFFE18EAF3CBB@MOBILEE1705>
	<B7667045-D5EF-4885-8548-BF7788053E3B@avgs.ca>
Message-ID: <m2w87f2f3b91005072250pe2de05c0nf83c69af2c80a7d5@mail.gmail.com>

It has its own page:
http://wiki.freeswitch.org/wiki/Variable_signal_bond

However it wasn't on the main chan vars page until a minute ago. ;)

-MC

On Fri, May 7, 2010 at 5:34 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> signal_bond
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 2010-05-07, at 8:32 PM, Madovsky wrote:
>
> Is there an already exsting variable which represent
> the UUID of B leg from the A leg phone once the call is established ?
> (sorry Mathieu I lost all my notes of this subject yesterday)
>
> Thanks
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From janvb at live.com  Fri May  7 23:47:59 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 8 May 2010 08:47:59 +0200
Subject: [Freeswitch-users] Alarms-codes
In-Reply-To: <z2o87f2f3b91005072246y39206088g6900a8a81434626c@mail.gmail.com>
References: <017a01caec8a$ee74bb70$cb5e3250$@com>,
	<BE3F6D4F-9486-43F3-ACAF-70C84CB001FF@gmail.com>,
	<01b101caec94$1f009fd0$5d01df70$@com>,
	<w2x87f2f3b91005051355j7c5c775dld43c79c6bf9d61cb@mail.gmail.com>,
	<A60A2CFD3B5C41D68FB970C1EB64DFF2@bp1.ad.bp.com>,
	<g2n6b65470d1005070808y30d1dce7q1c9b4fec8cc8f058@mail.gmail.com>,
	<i2g87f2f3b91005071459i16d8bd2fr7fa57a6304d2c0dc@mail.gmail.com>,
	<h2k23f91031005071842t865d2765x595a7a2572f13638@mail.gmail.com>,
	<z2o87f2f3b91005072246y39206088g6900a8a81434626c@mail.gmail.com>
Message-ID: <SNT135-w5FF0C5B6D49D392DF23CDD2F70@phx.gbl>


hi,

 

I have playing around with FreeSWITCHf or a while and one of things I am looking for is unique Alarms-codes

 

FreeSwitch has error, warning, info etc on the logging - but the message print module name and line-number - it does not incude a proper, unique alarm-code. Using the combination of module-name and line-number is possible - but this changes from release to release.

 

The point with alarms is that they are forwarded into databases/applications/operating senters that monitor telecom equipmenet. The way this usually is done is that the forwarding mechanism filter on alarm-code/severity as we don't want to forward junk - but it's a requirement that we visualise those that affect business. 

 

---

 

It's doable to change logging so that every module insert a unique number - starting with 1 - this would make modulename + alarm-code an unique combination that would serve the purpose.

 

Any suggestions?

 

Jan

 

 
 		 	   		  
_________________________________________________________________
Hotmail: Trusted email with Microsoft?s powerful SPAM protection.
https://signup.live.com/signup.aspx?id=60969
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From bwibowo at gmail.com  Sat May  8 01:26:06 2010
From: bwibowo at gmail.com (budi wibowo)
Date: Sat, 8 May 2010 15:26:06 +0700
Subject: [Freeswitch-users] outgoing gateway
In-Reply-To: <y2z23f91031005071930p3f96278aj9f7db3a5e0df60aa@mail.gmail.com>
References: <l2o547ec34a1005061642y37cd9880tac6e3aef1da1be1@mail.gmail.com>
	<t2u94f7dfb11005062034wc952a56ev92417d4456cc8b55@mail.gmail.com>
	<1077559925-1273204385-cardhu_decombobulator_blackberry.rim.net-1446324892-@bda057.bisx.prodap.on.blackberry>
	<472BC89E-11F0-49C2-8EA5-23783BC38770@jerris.com>
	<z2y547ec34a1005071904mcc2aa5d6wbec359c34d639f61@mail.gmail.com>
	<t2t23f91031005071926l451137ffo76d667030f41f573@mail.gmail.com>
	<y2z23f91031005071930p3f96278aj9f7db3a5e0df60aa@mail.gmail.com>
Message-ID: <r2h547ec34a1005080126i9e6f0cche81e1c1118ece6ef@mail.gmail.com>

dear all i try to make some changes but my call still failing
why the call never reach gateway i define in sip_profile/external/ and call
goes to voiprakyar.or.id that i never define




2010-05-08 15:22:46.403733 [NOTICE] switch_channel.c:602 New Channel
sofia/internal/budi at sip1.xxx.com [7e483bd8-ba9d-4f1e-8560-74edb8fba2af]
2010-05-08 15:22:46.407056 [INFO] mod_dialplan_xml.c:315 Processing
budi->62815145150 in context default
2010-05-08 15:22:46.410565 [NOTICE] switch_ivr.c:1349 Transfer
sofia/internal/budi at sip1.xxx.com to enum[62815145150 at default]
2010-05-08 15:22:46.750534 [NOTICE] switch_channel.c:602 New Channel
sofia/internal/62815145150 at voiprakyat.or.id[1e55a5c7-5d58-4437-a48b-4278154b57a0]
2010-05-08 15:22:48.426621 [NOTICE] sofia.c:3849 Hangup sofia/internal/
62815145150 at voiprakyat.or.id [CS_CONSUME_MEDIA] [CALL_REJECTED]


TIA

budi

On Sat, May 8, 2010 at 9:30 AM, Seven Du <dujinfang at gmail.com> wrote:

> In the mean time, I think it's a good article for you:
>
> http://www.linuxpromagazine.com/Issues/2009/106/TALK-SOFT
>
> 2010/5/8 Seven Du <dujinfang at gmail.com>:
> > default configuration files of mod_sofia is in conf/sip_profiles
> >
> > http://wiki.freeswitch.org/wiki/Sofia
> > http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files
> >
> > 2010/5/8 budi wibowo <bwibowo at gmail.com>:
> >> thx, after reading the document i still got confusion about sofia module
> >> i give example below
> >>
> >> Dialing Through A Gateway(SIP Provider)
> >>
> >> A gateway is a means for making outbound calls through a SIP provider.
> For
> >> example:
> >>
> >> sofia/gateway/mygateway.com/1234
> >>
> >> by default i dont find any directory named sofia, where i should put
> this
> >> directory?
> >> many document i read telling about sofia
> >> TIA
> >> budi
> >>
> >> On Fri, May 7, 2010 at 12:55 PM, Michael Jerris <mike at jerris.com>
> wrote:
> >>>
> >>> http://wiki.freeswitch.org/wiki/Dialplan
> >>> On May 6, 2010, at 11:53 PM, Budi wibowo wrote:
> >>>
> >>> Yes I have siphit installed, I tried to make some changes on dialplan
> file
> >>> but call always goes to other server, what should I configure to
> implement
> >>> this outgoing call
> >>>
> >>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> >
> >
> > --
> > Blog: http://www.dujinfang.com
> > Proj:  http://www.freeswitch.org.cn
> >
>
>
>
> --
> Blog: http://www.dujinfang.com
> Proj:  http://www.freeswitch.org.cn
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dujinfang at gmail.com  Sat May  8 01:41:59 2010
From: dujinfang at gmail.com (Seven Du)
Date: Sat, 8 May 2010 16:41:59 +0800
Subject: [Freeswitch-users] outgoing gateway
In-Reply-To: <r2h547ec34a1005080126i9e6f0cche81e1c1118ece6ef@mail.gmail.com>
References: <l2o547ec34a1005061642y37cd9880tac6e3aef1da1be1@mail.gmail.com>
	<t2u94f7dfb11005062034wc952a56ev92417d4456cc8b55@mail.gmail.com>
	<1077559925-1273204385-cardhu_decombobulator_blackberry.rim.net-1446324892-@bda057.bisx.prodap.on.blackberry>
	<472BC89E-11F0-49C2-8EA5-23783BC38770@jerris.com>
	<z2y547ec34a1005071904mcc2aa5d6wbec359c34d639f61@mail.gmail.com>
	<t2t23f91031005071926l451137ffo76d667030f41f573@mail.gmail.com>
	<y2z23f91031005071930p3f96278aj9f7db3a5e0df60aa@mail.gmail.com>
	<r2h547ec34a1005080126i9e6f0cche81e1c1118ece6ef@mail.gmail.com>
Message-ID: <z2n23f91031005080141pd9d99f02t8ff30f844beb8877@mail.gmail.com>

That won't help if you don't put a full log and your conf on pastebin.

2010/5/8 budi wibowo <bwibowo at gmail.com>:
> dear all i try to make some changes but my call still failing
> why the call never reach gateway i define in sip_profile/external/ and call
> goes to voiprakyar.or.id that i never define
>
>
>
> 2010-05-08 15:22:46.403733 [NOTICE] switch_channel.c:602 New Channel
> sofia/internal/budi at sip1.xxx.com [7e483bd8-ba9d-4f1e-8560-74edb8fba2af]
> 2010-05-08 15:22:46.407056 [INFO] mod_dialplan_xml.c:315 Processing
> budi->62815145150 in context default
> 2010-05-08 15:22:46.410565 [NOTICE] switch_ivr.c:1349 Transfer
> sofia/internal/budi at sip1.xxx.com to enum[62815145150 at default]
> 2010-05-08 15:22:46.750534 [NOTICE] switch_channel.c:602 New Channel
> sofia/internal/62815145150 at voiprakyat.or.id
> [1e55a5c7-5d58-4437-a48b-4278154b57a0]
> 2010-05-08 15:22:48.426621 [NOTICE] sofia.c:3849 Hangup
> sofia/internal/62815145150 at voiprakyat.or.id [CS_CONSUME_MEDIA]
> [CALL_REJECTED]
>
> TIA
> budi
> On Sat, May 8, 2010 at 9:30 AM, Seven Du <dujinfang at gmail.com> wrote:
>>
>> In the mean time, I think it's a good article for you:
>>
>> http://www.linuxpromagazine.com/Issues/2009/106/TALK-SOFT
>>
>> 2010/5/8 Seven Du <dujinfang at gmail.com>:
>> > default configuration files of mod_sofia is in conf/sip_profiles
>> >
>> > http://wiki.freeswitch.org/wiki/Sofia
>> > http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files
>> >
>> > 2010/5/8 budi wibowo <bwibowo at gmail.com>:
>> >> thx, after reading the document i still got confusion about sofia
>> >> module
>> >> i give example below
>> >>
>> >> Dialing Through A Gateway(SIP Provider)
>> >>
>> >> A gateway is a means for making outbound calls through a SIP provider.
>> >> For
>> >> example:
>> >>
>> >> sofia/gateway/mygateway.com/1234
>> >>
>> >> by default i dont find any directory named sofia, where i should put
>> >> this
>> >> directory?
>> >> many document i read telling about sofia
>> >> TIA
>> >> budi
>> >>
>> >> On Fri, May 7, 2010 at 12:55 PM, Michael Jerris <mike at jerris.com>
>> >> wrote:
>> >>>
>> >>> http://wiki.freeswitch.org/wiki/Dialplan
>> >>> On May 6, 2010, at 11:53 PM, Budi wibowo wrote:
>> >>>
>> >>> Yes I have siphit installed, I tried to make some changes on dialplan
>> >>> file
>> >>> but call always goes to other server, what should I configure to
>> >>> implement
>> >>> this outgoing call
>> >>>
>> >>>
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>>
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >
>> >
>> >
>> > --
>> > Blog: http://www.dujinfang.com
>> > Proj: ?http://www.freeswitch.org.cn
>> >
>>
>>
>>
>> --
>> Blog: http://www.dujinfang.com
>> Proj: ?http://www.freeswitch.org.cn
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



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


From david.ponzone at gmail.com  Sat May  8 02:04:55 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Sat, 8 May 2010 11:04:55 +0200
Subject: [Freeswitch-users] outgoing gateway
In-Reply-To: <r2h547ec34a1005080126i9e6f0cche81e1c1118ece6ef@mail.gmail.com>
References: <l2o547ec34a1005061642y37cd9880tac6e3aef1da1be1@mail.gmail.com>
	<t2u94f7dfb11005062034wc952a56ev92417d4456cc8b55@mail.gmail.com>
	<1077559925-1273204385-cardhu_decombobulator_blackberry.rim.net-1446324892-@bda057.bisx.prodap.on.blackberry>
	<472BC89E-11F0-49C2-8EA5-23783BC38770@jerris.com>
	<z2y547ec34a1005071904mcc2aa5d6wbec359c34d639f61@mail.gmail.com>
	<t2t23f91031005071926l451137ffo76d667030f41f573@mail.gmail.com>
	<y2z23f91031005071930p3f96278aj9f7db3a5e0df60aa@mail.gmail.com>
	<r2h547ec34a1005080126i9e6f0cche81e1c1118ece6ef@mail.gmail.com>
Message-ID: <09D47C25-81C0-4815-B477-7595D769EF60@gmail.com>

Not sure because your logs are toon short, but it sems that your call  
is routed to ENUM.
In the default config, ENUM routing is the last resort rule.
You need to add your own stuff before it.
To do so, you may add your extensions in a xml file called  
01_something.xml in conf/dialplan/default
(see 01_example.com.xml for an example).

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 08/05/2010 ? 10:26, budi wibowo a ?crit :

> dear all i try to make some changes but my call still failing
> why the call never reach gateway i define in sip_profile/external/  
> and call goes to voiprakyar.or.id that i never define
>
>
>
>
> 2010-05-08 15:22:46.403733 [NOTICE] switch_channel.c:602 New Channel sofia/internal/budi at sip1.xxx.com 
>  [7e483bd8-ba9d-4f1e-8560-74edb8fba2af]
> 2010-05-08 15:22:46.407056 [INFO] mod_dialplan_xml.c:315 Processing  
> budi->62815145150 in context default
> 2010-05-08 15:22:46.410565 [NOTICE] switch_ivr.c:1349 Transfer sofia/internal/budi at sip1.xxx.com 
>  to enum[62815145150 at default]
> 2010-05-08 15:22:46.750534 [NOTICE] switch_channel.c:602 New Channel sofia/internal/62815145150 at voiprakyat.or.id 
>  [1e55a5c7-5d58-4437-a48b-4278154b57a0]
> 2010-05-08 15:22:48.426621 [NOTICE] sofia.c:3849 Hangup sofia/internal/62815145150 at voiprakyat.or.id 
>  [CS_CONSUME_MEDIA] [CALL_REJECTED]
>
>
> TIA
>
> budi
>
> On Sat, May 8, 2010 at 9:30 AM, Seven Du <dujinfang at gmail.com> wrote:
> In the mean time, I think it's a good article for you:
>
> http://www.linuxpromagazine.com/Issues/2009/106/TALK-SOFT
>
> 2010/5/8 Seven Du <dujinfang at gmail.com>:
> > default configuration files of mod_sofia is in conf/sip_profiles
> >
> > http://wiki.freeswitch.org/wiki/Sofia
> > http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files
> >
> > 2010/5/8 budi wibowo <bwibowo at gmail.com>:
> >> thx, after reading the document i still got confusion about sofia  
> module
> >> i give example below
> >>
> >> Dialing Through A Gateway(SIP Provider)
> >>
> >> A gateway is a means for making outbound calls through a SIP  
> provider. For
> >> example:
> >>
> >> sofia/gateway/mygateway.com/1234
> >>
> >> by default i dont find any directory named sofia, where i should  
> put this
> >> directory?
> >> many document i read telling about sofia
> >> TIA
> >> budi
> >>
> >> On Fri, May 7, 2010 at 12:55 PM, Michael Jerris <mike at jerris.com>  
> wrote:
> >>>
> >>> http://wiki.freeswitch.org/wiki/Dialplan
> >>> On May 6, 2010, at 11:53 PM, Budi wibowo wrote:
> >>>
> >>> Yes I have siphit installed, I tried to make some changes on  
> dialplan file
> >>> but call always goes to other server, what should I configure to  
> implement
> >>> this outgoing call
> >>>
> >>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> >
> >
> > --
> > Blog: http://www.dujinfang.com
> > Proj:  http://www.freeswitch.org.cn
> >
>
>
>
> --
> Blog: http://www.dujinfang.com
> Proj:  http://www.freeswitch.org.cn
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From babak.freeswitch at gmail.com  Sat May  8 03:50:01 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Sat, 8 May 2010 15:20:01 +0430
Subject: [Freeswitch-users] voicemail recording problem
Message-ID: <AANLkTimOT5Jsw4ccNCzGzkVd9CiXGIIwHyTcpqJkWiCs@mail.gmail.com>

Hi
I'm using the below ivr menu  and sending the caller to voice mail by
transferring to an extension but the problem is that beep tone is also
recorded!
I mean the recorded voice mails begin with beep sound (deeeed)!!!!
thanks
<menu name="sepanta_ivr"
      greet-long="sepanta/mainMenu.wav"
      invalid-sound="sepanta/digitCollectError.wav"
      confirm-key=""
      confirm-attempts="3"
      timeout="10000"
      inter-digit-timeout="2000"
      max-failures="3"
      max-timeouts="3"
      digit-len="1">
    <entry action="menu-exec-app" digits="1" param="managed Answering 1"/> <!--
ADSL sale service  -->
    <entry action="menu-exec-app" digits="2" param="managed Answering 2"/>
 <!-- Customer services and contract renewal  -->
* **<entry action="menu-exec-app" digits="3" param="transfer
manager-voice-mail"/>    <!-- Managing Director -->*
    <entry action="menu-sub" digits="4" param="sepanta_ivr_submenu"/>


     <extension name="manager-voice-mail">
<condition field="destination_number" expression="^manager-voice-mail$">
<action application="set" data="voicemail_greeting_number=2"/>
<action application="set" data="skip_instructions=true"/>
        <action application="voicemail" data="default ${domain_name}
kiarash"/>
      </condition>
    </extension>
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From janvb at live.com  Sat May  8 03:51:39 2010
From: janvb at live.com (Jan Berger)
Date: Sat, 8 May 2010 12:51:39 +0200
Subject: [Freeswitch-users] C#/EventSocket
In-Reply-To: <09D47C25-81C0-4815-B477-7595D769EF60@gmail.com>
References: <l2o547ec34a1005061642y37cd9880tac6e3aef1da1be1@mail.gmail.com>,
	<t2u94f7dfb11005062034wc952a56ev92417d4456cc8b55@mail.gmail.com>,
	<1077559925-1273204385-cardhu_decombobulator_blackberry.rim.net-1446324892-@bda057.bisx.prodap.on.blackberry>,
	<472BC89E-11F0-49C2-8EA5-23783BC38770@jerris.com>,
	<z2y547ec34a1005071904mcc2aa5d6wbec359c34d639f61@mail.gmail.com>,
	<t2t23f91031005071926l451137ffo76d667030f41f573@mail.gmail.com>,
	<y2z23f91031005071930p3f96278aj9f7db3a5e0df60aa@mail.gmail.com>,
	<r2h547ec34a1005080126i9e6f0cche81e1c1118ece6ef@mail.gmail.com>,
	<09D47C25-81C0-4815-B477-7595D769EF60@gmail.com>
Message-ID: <SNT135-w419FCF0C86467A2CDA3812D2F70@phx.gbl>


hi,

 

I am looking into event_socket and the C# demo in contributions, but I get a request for a password when I try to use this?

 

EventSocket.pfx

 

Jan
 		 	   		  
_________________________________________________________________
Hotmail: Free, trusted and rich email service.
https://signup.live.com/signup.aspx?id=60969
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From tayeb.meftah at gmail.com  Sun May  9 05:21:18 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sun, 09 May 2010 14:21:18 +0200
Subject: [Freeswitch-users] call privacy
Message-ID: <4BE6A8BE.1060104@gmail.com>

hello
how do i let user do a hiden call?
using privacy tools?
and, if i get a hidden call and my freeswitch is a node in our telecom 
infrastructure, how do i force the caller id number to display?
note, i am not a customer but a core switch
thanks


From david.ponzone at gmail.com  Sat May  8 06:37:12 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Sat, 8 May 2010 15:37:12 +0200
Subject: [Freeswitch-users] call privacy
In-Reply-To: <4BE6A8BE.1060104@gmail.com>
References: <4BE6A8BE.1060104@gmail.com>
Message-ID: <E2DD62A0-5FF7-4281-9677-46565FEC0E1E@gmail.com>

As I told you on IRC, there is no standard way to do that.

All depends on how the upstream switch sends you the informations you  
want to preserve.
You have to check the chanvars you get on leg A, and manipulation the  
header of leg B accordingly.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 09/05/2010 ? 14:21, Meftah Tayeb a ?crit :

> hello
> how do i let user do a hiden call?
> using privacy tools?
> and, if i get a hidden call and my freeswitch is a node in our telecom
> infrastructure, how do i force the caller id number to display?
> note, i am not a customer but a core switch
> thanks
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From dftoro at yahoo.com  Sat May  8 09:32:35 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Sat, 8 May 2010 09:32:35 -0700 (PDT)
Subject: [Freeswitch-users] C#/EventSocket
In-Reply-To: <SNT135-w419FCF0C86467A2CDA3812D2F70@phx.gbl>
Message-ID: <514780.42671.qm@web33506.mail.mud.yahoo.com>

Hi, check libs\esl\managed

Diego Toro

http://lacarretade.blogspot.com/

--- On Sat, 5/8/10, Jan Berger <janvb at live.com> wrote:

From: Jan Berger <janvb at live.com>
Subject: [Freeswitch-users] C#/EventSocket
To: freeswitch-users at lists.freeswitch.org
Date: Saturday, May 8, 2010, 5:51 AM




hi,

?

I am looking into event_socket and the C# demo in contributions, but I get a request for a password when I try to use this?

?

EventSocket.pfx

?

Jan
 		 	   		  
Hotmail: Free, trusted and rich email service. Get it now. 

-----Inline Attachment Follows-----

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



      
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From dftoro at yahoo.com  Sat May  8 09:44:32 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Sat, 8 May 2010 09:44:32 -0700 (PDT)
Subject: [Freeswitch-users] Sound files French, German, and Spanish
Message-ID: <981664.2998.qm@web33505.mail.mud.yahoo.com>

Hi all,

The lastest announcement the official release of version 1.0.6, it says, 
Sound files have been created for French, German, and Spanish.

Where Can I get these sound files ?

Thank you

Diego Toro
http://lacarretade.blogspot.com/


      


From babak.freeswitch at gmail.com  Sat May  8 10:55:48 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Sat, 8 May 2010 22:25:48 +0430
Subject: [Freeswitch-users] voicemail recording problem
In-Reply-To: <AANLkTimOT5Jsw4ccNCzGzkVd9CiXGIIwHyTcpqJkWiCs@mail.gmail.com>
References: <AANLkTimOT5Jsw4ccNCzGzkVd9CiXGIIwHyTcpqJkWiCs@mail.gmail.com>
Message-ID: <AANLkTimAKsVsiCmL37OZ-rqsgEmUP2uSyFkBH2OcJaue@mail.gmail.com>

sorry for this silly question!
the reason is I was testing on my laptop which has a built in mic so the
tone was recorded!
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From moises.silva at gmail.com  Sat May  8 11:01:05 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Sat, 8 May 2010 14:01:05 -0400
Subject: [Freeswitch-users] Alarms-codes
In-Reply-To: <SNT135-w5FF0C5B6D49D392DF23CDD2F70@phx.gbl>
References: <017a01caec8a$ee74bb70$cb5e3250$@com>
	<BE3F6D4F-9486-43F3-ACAF-70C84CB001FF@gmail.com>
	<01b101caec94$1f009fd0$5d01df70$@com>
	<w2x87f2f3b91005051355j7c5c775dld43c79c6bf9d61cb@mail.gmail.com>
	<A60A2CFD3B5C41D68FB970C1EB64DFF2@bp1.ad.bp.com>
	<g2n6b65470d1005070808y30d1dce7q1c9b4fec8cc8f058@mail.gmail.com>
	<i2g87f2f3b91005071459i16d8bd2fr7fa57a6304d2c0dc@mail.gmail.com>
	<h2k23f91031005071842t865d2765x595a7a2572f13638@mail.gmail.com>
	<z2o87f2f3b91005072246y39206088g6900a8a81434626c@mail.gmail.com>
	<SNT135-w5FF0C5B6D49D392DF23CDD2F70@phx.gbl>
Message-ID: <y2zc4d05cbe1005081101ma76fb71do61a7469d1990096c@mail.gmail.com>

Hello Jan,

I believe you're confusing alarms with logging and those two are not the
same thing and I think should not be mixed. For alarms an special event
should be triggered ( as mentioned in other thread, may be SWITCH_EVENT_TRAP
).

mod_freetdm ( FreeSWITCH PSTN module ) is already sending a
SWITCH_EVENT_TRAP when any pstn port becomes alarmed, other modules may
start using it for their own type of alarms. Then we may add the alarm-code
and severity fields.

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com


On Sat, May 8, 2010 at 2:47 AM, Jan Berger <janvb at live.com> wrote:

>  hi,
>
> I have playing around with FreeSWITCHf or a while and one of things I am
> looking for is unique Alarms-codes
>
> FreeSwitch has error, warning, info etc on the logging - but the message
> print module name and line-number - it does not incude a proper, unique
> alarm-code. Using the combination of module-name and line-number is possible
> - but this changes from release to release.
>
> The point with alarms is that they are forwarded into
> databases/applications/operating senters that monitor telecom equipmenet.
> The way this usually is done is that the forwarding mechanism filter on
> alarm-code/severity as we don't want to forward junk - but it's a
> requirement that we visualise those that affect business.
>
> ---
>
> It's doable to change logging so that every module insert a unique number -
> starting with 1 - this would make modulename + alarm-code an unique
> combination that would serve the purpose.
>
> Any suggestions?
>
> Jan
>
>
>
> ------------------------------
> Hotmail: Trusted email with Microsoft?s powerful SPAM protection. Sign up
> now. <https://signup.live.com/signup.aspx?id=60969>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Sat May  8 11:56:29 2010
From: msc at freeswitch.org (Michael S Collins)
Date: Sat, 8 May 2010 11:56:29 -0700
Subject: [Freeswitch-users] Sound files French, German, and Spanish
In-Reply-To: <981664.2998.qm@web33505.mail.mud.yahoo.com>
References: <981664.2998.qm@web33505.mail.mud.yahoo.com>
Message-ID: <156D94E4-5A23-4D26-A5D5-23BF30AEE23D@freeswitch.org>

Diego,
Our intrepid users have made sound files in these languages. The  
French and German files were mentioned in the mailing list so search  
the archives. We found some missing sounds in the Spanish sound set an  
we are reviewing them. The Spanish sounds are Latin American. We also  
have a bounty for Castilian sounds.

-MC

Sent from my iPhone

On May 8, 2010, at 9:44 AM, Diego Toro <dftoro at yahoo.com> wrote:

> Hi all,
>
> The lastest announcement the official release of version 1.0.6, it  
> says,
> Sound files have been created for French, German, and Spanish.
>
> Where Can I get these sound files ?
>
> Thank you
>
> Diego Toro
> http://lacarretade.blogspot.com/
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From infos at madovsky.org  Sat May  8 14:07:45 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 8 May 2010 17:07:45 -0400
Subject: [Freeswitch-users] ivr codec
Message-ID: <A4DE2B833B9049D5A9F39D4D0910F588@MOBILEE1705>

I just noticed in the log this :

switch_ivr_play_say.c:1152 Codec Activated L16 at 16000hz 1 channels 20ms

when voicemail is answering. But this codec is not in vars.xml list.
why this codec is activated with IVR ?

Thanks

Franck
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From brian at freeswitch.org  Sat May  8 14:16:17 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 8 May 2010 16:16:17 -0500
Subject: [Freeswitch-users] ivr codec
In-Reply-To: <A4DE2B833B9049D5A9F39D4D0910F588@MOBILEE1705>
References: <A4DE2B833B9049D5A9F39D4D0910F588@MOBILEE1705>
Message-ID: <E0379CA5-3293-4E1D-955E-E7440520A44F@freeswitch.org>

Because the files are recorded in L16 wav files.

/b

On May 8, 2010, at 4:07 PM, Madovsky wrote:

> I just noticed in the log this :
>  
> switch_ivr_play_say.c:1152 Codec Activated L16 at 16000hz 1 channels 20ms
>  
> when voicemail is answering. But this codec is not in vars.xml list.
> why this codec is activated with IVR ?
>  
> Thanks
> 
> Franck

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From infos at madovsky.org  Sat May  8 14:48:16 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 8 May 2010 17:48:16 -0400
Subject: [Freeswitch-users] ivr codec
References: <A4DE2B833B9049D5A9F39D4D0910F588@MOBILEE1705>
	<E0379CA5-3293-4E1D-955E-E7440520A44F@freeswitch.org>
Message-ID: <D3041DEA8E394EA190C3B518C230937D@MOBILEE1705>

ok didn't know L16 was .wav codec.

thx
  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Saturday, May 08, 2010 5:16 PM
  Subject: Re: [Freeswitch-users] ivr codec


  Because the files are recorded in L16 wav files.


  /b


  On May 8, 2010, at 4:07 PM, Madovsky wrote:


    I just noticed in the log this :

    switch_ivr_play_say.c:1152 Codec Activated L16 at 16000hz 1 channels 20ms

    when voicemail is answering. But this codec is not in vars.xml list.
    why this codec is activated with IVR ?

    Thanks

    Franck




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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From brian at freeswitch.org  Sat May  8 15:04:41 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 8 May 2010 17:04:41 -0500
Subject: [Freeswitch-users] ivr codec
In-Reply-To: <D3041DEA8E394EA190C3B518C230937D@MOBILEE1705>
References: <A4DE2B833B9049D5A9F39D4D0910F588@MOBILEE1705>
	<E0379CA5-3293-4E1D-955E-E7440520A44F@freeswitch.org>
	<D3041DEA8E394EA190C3B518C230937D@MOBILEE1705>
Message-ID: <59A6D525-BEA7-47DB-A0BB-2AF228BB3973@freeswitch.org>

No wav is a container its storing L16 into the wav file.

/b

On May 8, 2010, at 4:48 PM, Madovsky wrote:

> ok didn't know L16 was .wav codec.
>  
> thx

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From bwibowo at gmail.com  Sat May  8 17:16:15 2010
From: bwibowo at gmail.com (budi wibowo)
Date: Sun, 9 May 2010 07:16:15 +0700
Subject: [Freeswitch-users] outgoing gateway
In-Reply-To: <09D47C25-81C0-4815-B477-7595D769EF60@gmail.com>
References: <l2o547ec34a1005061642y37cd9880tac6e3aef1da1be1@mail.gmail.com>
	<t2u94f7dfb11005062034wc952a56ev92417d4456cc8b55@mail.gmail.com>
	<1077559925-1273204385-cardhu_decombobulator_blackberry.rim.net-1446324892-@bda057.bisx.prodap.on.blackberry>
	<472BC89E-11F0-49C2-8EA5-23783BC38770@jerris.com>
	<z2y547ec34a1005071904mcc2aa5d6wbec359c34d639f61@mail.gmail.com>
	<t2t23f91031005071926l451137ffo76d667030f41f573@mail.gmail.com>
	<y2z23f91031005071930p3f96278aj9f7db3a5e0df60aa@mail.gmail.com>
	<r2h547ec34a1005080126i9e6f0cche81e1c1118ece6ef@mail.gmail.com>
	<09D47C25-81C0-4815-B477-7595D769EF60@gmail.com>
Message-ID: <h2s547ec34a1005081716xc09243cq20ec07fda9ba863c@mail.gmail.com>

thx a lot for your explanation, i have another question, my sip server (mera
sip-hit) dont need user name and password to connect. for security i just
use firewall rules.
if i refer to other samples like fwd,voicheap etc all require user name and
password.

my conf/sip_profiles/external/test.xml look like this
 <include>
        <gateway name="test">
          <param name="realm" value="202.xx.xx.xx.xx"/>
          <param name="proxy" value="202.xx.xx.xx.xx"/>
          <param name="expire-seconds" value="3600"/>
          <param name="register" value="true"/>
          <param name="retry-seconds" value="3600"/>
        </gateway>
</include>


my conf/dialplan/default/00_test.xml look like

<include>
   <extension name="test">
     <condition field="destination_number" expression="^(\d)$">
       <action application="set"
data="effective_caller_id_number=${outbound_caller_id_number}"/>
       <action application="set"
data="effective_caller_id_name=${outbound_caller_id_name}"/>
       <action application="bridge" data="sofia/gateway/test/$1"/>
     </condition>
   </extension>
</include>


but still i cant see any call coming from my mera siphit.

TIA

budi wibowo



On Sat, May 8, 2010 at 4:04 PM, David Ponzone <david.ponzone at gmail.com>wrote:

> Not sure because your logs are toon short, but it sems that your call is
> routed to ENUM.
> In the default config, ENUM routing is the last resort rule.
> You need to add your own stuff before it.
> To do so, you may add your extensions in a xml file called 01_something.xml
> in conf/dialplan/default
> (see 01_example.com.xml for an example).
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 08/05/2010 ? 10:26, budi wibowo a ?crit :
>
> dear all i try to make some changes but my call still failing
> why the call never reach gateway i define in sip_profile/external/ and call
> goes to voiprakyar.or.id that i never define
>
>
>
>
> 2010-05-08 15:22:46.403733 [NOTICE] switch_channel.c:602 New Channel
> sofia/internal/budi at sip1.xxx.com [7e483bd8-ba9d-4f1e-8560-74edb8fba2af]
> 2010-05-08 15:22:46.407056 [INFO] mod_dialplan_xml.c:315 Processing
> budi->62815145150 in context default
> 2010-05-08 15:22:46.410565 [NOTICE] switch_ivr.c:1349 Transfer
> sofia/internal/budi at sip1.xxx.com to enum[62815145150 at default]
> 2010-05-08 15:22:46.750534 [NOTICE] switch_channel.c:602 New Channel
> sofia/internal/62815145150 at voiprakyat.or.id[1e55a5c7-5d58-4437-a48b-4278154b57a0]
> 2010-05-08 15:22:48.426621 [NOTICE] sofia.c:3849 Hangup sofia/internal/
> 62815145150 at voiprakyat.or.id [CS_CONSUME_MEDIA] [CALL_REJECTED]
>
>
> TIA
>
> budi
>
> On Sat, May 8, 2010 at 9:30 AM, Seven Du <dujinfang at gmail.com> wrote:
>
>> In the mean time, I think it's a good article for you:
>>
>> http://www.linuxpromagazine.com/Issues/2009/106/TALK-SOFT
>>
>> 2010/5/8 Seven Du <dujinfang at gmail.com>:
>> > default configuration files of mod_sofia is in conf/sip_profiles
>> >
>> > http://wiki.freeswitch.org/wiki/Sofia
>> > http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files
>> >
>> > 2010/5/8 budi wibowo <bwibowo at gmail.com>:
>> >> thx, after reading the document i still got confusion about sofia
>> module
>> >> i give example below
>> >>
>> >> Dialing Through A Gateway(SIP Provider)
>> >>
>> >> A gateway is a means for making outbound calls through a SIP provider.
>> For
>> >> example:
>> >>
>> >> sofia/gateway/mygateway.com/1234
>> >>
>> >> by default i dont find any directory named sofia, where i should put
>> this
>> >> directory?
>> >> many document i read telling about sofia
>> >> TIA
>> >> budi
>> >>
>> >> On Fri, May 7, 2010 at 12:55 PM, Michael Jerris <mike at jerris.com>
>> wrote:
>> >>>
>> >>> http://wiki.freeswitch.org/wiki/Dialplan
>> >>> On May 6, 2010, at 11:53 PM, Budi wibowo wrote:
>> >>>
>> >>> Yes I have siphit installed, I tried to make some changes on dialplan
>> file
>> >>> but call always goes to other server, what should I configure to
>> implement
>> >>> this outgoing call
>> >>>
>> >>>
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>>
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >
>> >
>> >
>> > --
>> > Blog: http://www.dujinfang.com
>> > Proj:  http://www.freeswitch.org.cn
>> >
>>
>>
>>
>> --
>> Blog: http://www.dujinfang.com
>> Proj:  http://www.freeswitch.org.cn
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Sat May  8 20:25:42 2010
From: infos at madovsky.org (Madovsky)
Date: Sat, 8 May 2010 23:25:42 -0400
Subject: [Freeswitch-users] signal_bond and uuid
Message-ID: <2A08815A59134650A928BAC2C6417D58@MOBILEE1705>

if for example I d like to send dtmf to the legB in api mod_commands
should I use signal_bond as uuid of the channel ?

Thanks

F
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From infos at madovsky.org  Sat May  8 21:05:32 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 9 May 2010 00:05:32 -0400
Subject: [Freeswitch-users] uuid_send_dtmf question
Message-ID: <25835C58F30141559098DCA71D136606@MOBILEE1705>

Hi,

if I use call-id as uuid and the call-id has a form of 332434545 at domain.ltd

and I send dtmf with uuid_send_dtmf to the TO (default)
so I got this error from the FROM leg :

Invalid RTP parse error 'NoneType' object has no attribute 'name'

any idea ?

Thanks

F
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From dujinfang at gmail.com  Sun May  9 00:07:13 2010
From: dujinfang at gmail.com (Seven Du)
Date: Sun, 9 May 2010 15:07:13 +0800
Subject: [Freeswitch-users] outgoing gateway
In-Reply-To: <h2s547ec34a1005081716xc09243cq20ec07fda9ba863c@mail.gmail.com>
References: <l2o547ec34a1005061642y37cd9880tac6e3aef1da1be1@mail.gmail.com>
	<t2u94f7dfb11005062034wc952a56ev92417d4456cc8b55@mail.gmail.com>
	<1077559925-1273204385-cardhu_decombobulator_blackberry.rim.net-1446324892-@bda057.bisx.prodap.on.blackberry>
	<472BC89E-11F0-49C2-8EA5-23783BC38770@jerris.com>
	<z2y547ec34a1005071904mcc2aa5d6wbec359c34d639f61@mail.gmail.com>
	<t2t23f91031005071926l451137ffo76d667030f41f573@mail.gmail.com>
	<y2z23f91031005071930p3f96278aj9f7db3a5e0df60aa@mail.gmail.com>
	<r2h547ec34a1005080126i9e6f0cche81e1c1118ece6ef@mail.gmail.com>
	<09D47C25-81C0-4815-B477-7595D769EF60@gmail.com>
	<h2s547ec34a1005081716xc09243cq20ec07fda9ba863c@mail.gmail.com>
Message-ID: <m2m23f91031005090007h83fadcd6w744b3f4542322c13@mail.gmail.com>

again, pastebin your freeswitch logs so others can help you. don't
forget to press F8 before capture your logs.

2010/5/9 budi wibowo <bwibowo at gmail.com>:
> thx a lot for your explanation, i have another question, my sip server (mera
> sip-hit) dont need user name and password to connect. for security i just
> use firewall rules.
> if i refer to other samples like fwd,voicheap etc all require user name and
> password.
> my conf/sip_profiles/external/test.xml look like this
> ?<include>
> ?? ? ? ?<gateway name="test">
> ?? ? ? ? ?<param name="realm" value="202.xx.xx.xx.xx"/>
> ?? ? ? ? ?<param name="proxy" value="202.xx.xx.xx.xx"/>
> ?? ? ? ? ?<param name="expire-seconds" value="3600"/>
> ?? ? ? ? ?<param name="register" value="true"/>
> ?? ? ? ? ?<param name="retry-seconds" value="3600"/>
> ?? ? ? ?</gateway>
> </include>
>
> my conf/dialplan/default/00_test.xml look like
> <include>
> ?? <extension name="test">
> ?? ? <condition field="destination_number" expression="^(\d)$">
> ?? ? ? <action application="set"
> data="effective_caller_id_number=${outbound_caller_id_number}"/>
> ?? ? ? <action application="set"
> data="effective_caller_id_name=${outbound_caller_id_name}"/>
> ?? ? ? <action application="bridge" data="sofia/gateway/test/$1"/>
> ?? ? </condition>
> ?? </extension>
> </include>
>
> but still i cant see any call coming from my mera siphit.
> TIA
> budi wibowo
>
>
> On Sat, May 8, 2010 at 4:04 PM, David Ponzone <david.ponzone at gmail.com>
> wrote:
>>
>> Not sure because your logs are toon short, but it sems that your call is
>> routed to ENUM.
>> In the default config, ENUM routing is the last resort rule.
>> You need to add your own stuff before it.
>> To do so, you may add your extensions in a xml file called
>> 01_something.xml in conf/dialplan/default
>> (see 01_example.com.xml for an example).
>> David Ponzone ?Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel: ? ? ?01 74 03 18 97
>> gsm: ? 06 66 98 76 34
>> Service Client?IPeva
>> tel: ? ? ?0811 46 26 26
>> www.ipeva.fr? -? ?www.ipeva-studio.com
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>
>>
>>
>> Le 08/05/2010 ? 10:26, budi wibowo a ?crit :
>>
>> dear all i try to make some changes but my call still failing
>> why the call never reach gateway i define in sip_profile/external/ and
>> call goes to voiprakyar.or.id that i never define
>>
>>
>>
>> 2010-05-08 15:22:46.403733 [NOTICE] switch_channel.c:602 New Channel
>> sofia/internal/budi at sip1.xxx.com [7e483bd8-ba9d-4f1e-8560-74edb8fba2af]
>> 2010-05-08 15:22:46.407056 [INFO] mod_dialplan_xml.c:315 Processing
>> budi->62815145150 in context default
>> 2010-05-08 15:22:46.410565 [NOTICE] switch_ivr.c:1349 Transfer
>> sofia/internal/budi at sip1.xxx.com to enum[62815145150 at default]
>> 2010-05-08 15:22:46.750534 [NOTICE] switch_channel.c:602 New Channel
>> sofia/internal/62815145150 at voiprakyat.or.id
>> [1e55a5c7-5d58-4437-a48b-4278154b57a0]
>> 2010-05-08 15:22:48.426621 [NOTICE] sofia.c:3849 Hangup
>> sofia/internal/62815145150 at voiprakyat.or.id [CS_CONSUME_MEDIA]
>> [CALL_REJECTED]
>>
>> TIA
>> budi
>> On Sat, May 8, 2010 at 9:30 AM, Seven Du <dujinfang at gmail.com> wrote:
>>>
>>> In the mean time, I think it's a good article for you:
>>>
>>> http://www.linuxpromagazine.com/Issues/2009/106/TALK-SOFT
>>>
>>> 2010/5/8 Seven Du <dujinfang at gmail.com>:
>>> > default configuration files of mod_sofia is in conf/sip_profiles
>>> >
>>> > http://wiki.freeswitch.org/wiki/Sofia
>>> > http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files
>>> >
>>> > 2010/5/8 budi wibowo <bwibowo at gmail.com>:
>>> >> thx, after reading the document i still got confusion about sofia
>>> >> module
>>> >> i give example below
>>> >>
>>> >> Dialing Through A Gateway(SIP Provider)
>>> >>
>>> >> A gateway is a means for making outbound calls through a SIP provider.
>>> >> For
>>> >> example:
>>> >>
>>> >> sofia/gateway/mygateway.com/1234
>>> >>
>>> >> by default i dont find any directory named sofia, where i should put
>>> >> this
>>> >> directory?
>>> >> many document i read telling about sofia
>>> >> TIA
>>> >> budi
>>> >>
>>> >> On Fri, May 7, 2010 at 12:55 PM, Michael Jerris <mike at jerris.com>
>>> >> wrote:
>>> >>>
>>> >>> http://wiki.freeswitch.org/wiki/Dialplan
>>> >>> On May 6, 2010, at 11:53 PM, Budi wibowo wrote:
>>> >>>
>>> >>> Yes I have siphit installed, I tried to make some changes on dialplan
>>> >>> file
>>> >>> but call always goes to other server, what should I configure to
>>> >>> implement
>>> >>> this outgoing call
>>> >>>
>>> >>>
>>> >>>
>>> >>> _______________________________________________
>>> >>> FreeSWITCH-users mailing list
>>> >>> FreeSWITCH-users at lists.freeswitch.org
>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>
>>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>> http://www.freeswitch.org
>>> >>>
>>> >>
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >>
>>> >>
>>> >
>>> >
>>> >
>>> > --
>>> > Blog: http://www.dujinfang.com
>>> > Proj: ?http://www.freeswitch.org.cn
>>> >
>>>
>>>
>>>
>>> --
>>> Blog: http://www.dujinfang.com
>>> Proj: ?http://www.freeswitch.org.cn
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



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


From infos at madovsky.org  Sun May  9 01:25:17 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 9 May 2010 04:25:17 -0400
Subject: [Freeswitch-users] default voiceamail sentences
Message-ID: <AC78A683BB3B4B6B84755884FD08B792@MOBILEE1705>

I'd like to clean the default voicemail answer when user is not registered like
the person at extension xxxx is not available, record your message after the tone...
and once the message is registered say "saved, goodbye" only without "press 1 etc.."
possible ?

Thanks

Franck
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From david.ponzone at gmail.com  Sun May  9 02:49:08 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Sun, 9 May 2010 11:49:08 +0200
Subject: [Freeswitch-users] outgoing gateway
In-Reply-To: <h2s547ec34a1005081716xc09243cq20ec07fda9ba863c@mail.gmail.com>
References: <l2o547ec34a1005061642y37cd9880tac6e3aef1da1be1@mail.gmail.com>
	<t2u94f7dfb11005062034wc952a56ev92417d4456cc8b55@mail.gmail.com>
	<1077559925-1273204385-cardhu_decombobulator_blackberry.rim.net-1446324892-@bda057.bisx.prodap.on.blackberry>
	<472BC89E-11F0-49C2-8EA5-23783BC38770@jerris.com>
	<z2y547ec34a1005071904mcc2aa5d6wbec359c34d639f61@mail.gmail.com>
	<t2t23f91031005071926l451137ffo76d667030f41f573@mail.gmail.com>
	<y2z23f91031005071930p3f96278aj9f7db3a5e0df60aa@mail.gmail.com>
	<r2h547ec34a1005080126i9e6f0cche81e1c1118ece6ef@mail.gmail.com>
	<09D47C25-81C0-4815-B477-7595D769EF60@gmail.com>
	<h2s547ec34a1005081716xc09243cq20ec07fda9ba863c@mail.gmail.com>
Message-ID: <0EC75EB8-42A9-4C56-9DA7-27C8AA642131@gmail.com>

You need to add this in your gateway config:

   <param name="username" value="none"/>
   <param name="password" value="none"/>

and change the register param to:
           <param name="register" value="false"/>


David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 09/05/2010 ? 02:16, budi wibowo a ?crit :

> thx a lot for your explanation, i have another question, my sip  
> server (mera sip-hit) dont need user name and password to connect.  
> for security i just use firewall rules.
> if i refer to other samples like fwd,voicheap etc all require user  
> name and password.
>
> my conf/sip_profiles/external/test.xml look like this
>  <include>
>         <gateway name="test">
>           <param name="realm" value="202.xx.xx.xx.xx"/>
>           <param name="proxy" value="202.xx.xx.xx.xx"/>
>           <param name="expire-seconds" value="3600"/>
>           <param name="register" value="true"/>
>           <param name="retry-seconds" value="3600"/>
>         </gateway>
> </include>
>
>
> my conf/dialplan/default/00_test.xml look like
>
> <include>
>    <extension name="test">
>      <condition field="destination_number" expression="^(\d)$">
>        <action application="set" data="effective_caller_id_number=$ 
> {outbound_caller_id_number}"/>
>        <action application="set" data="effective_caller_id_name=$ 
> {outbound_caller_id_name}"/>
>        <action application="bridge" data="sofia/gateway/test/$1"/>
>      </condition>
>    </extension>
> </include>
>
>
> but still i cant see any call coming from my mera siphit.
>
> TIA
>
> budi wibowo
>
>
>
> On Sat, May 8, 2010 at 4:04 PM, David Ponzone  
> <david.ponzone at gmail.com> wrote:
> Not sure because your logs are toon short, but it sems that your  
> call is routed to ENUM.
> In the default config, ENUM routing is the last resort rule.
> You need to add your own stuff before it.
> To do so, you may add your extensions in a xml file called  
> 01_something.xml in conf/dialplan/default
> (see 01_example.com.xml for an example).
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et  
> ?tablis ? l'intention exclusive de ses destinataires. Toute  
> utilisation ou diffusion non autoris?e est interdite. Tout message  
> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou  
> falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le  
> d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
>
> Le 08/05/2010 ? 10:26, budi wibowo a ?crit :
>
>> dear all i try to make some changes but my call still failing
>> why the call never reach gateway i define in sip_profile/external/  
>> and call goes to voiprakyar.or.id that i never define
>>
>>
>>
>>
>> 2010-05-08 15:22:46.403733 [NOTICE] switch_channel.c:602 New  
>> Channel sofia/internal/budi at sip1.xxx.com [7e483bd8- 
>> ba9d-4f1e-8560-74edb8fba2af]
>> 2010-05-08 15:22:46.407056 [INFO] mod_dialplan_xml.c:315 Processing  
>> budi->62815145150 in context default
>> 2010-05-08 15:22:46.410565 [NOTICE] switch_ivr.c:1349 Transfer sofia/internal/budi at sip1.xxx.com 
>>  to enum[62815145150 at default]
>> 2010-05-08 15:22:46.750534 [NOTICE] switch_channel.c:602 New  
>> Channel sofia/internal/62815145150 at voiprakyat.or.id  
>> [1e55a5c7-5d58-4437-a48b-4278154b57a0]
>> 2010-05-08 15:22:48.426621 [NOTICE] sofia.c:3849 Hangup sofia/internal/62815145150 at voiprakyat.or.id 
>>  [CS_CONSUME_MEDIA] [CALL_REJECTED]
>>
>>
>> TIA
>>
>> budi
>>
>> On Sat, May 8, 2010 at 9:30 AM, Seven Du <dujinfang at gmail.com> wrote:
>> In the mean time, I think it's a good article for you:
>>
>> http://www.linuxpromagazine.com/Issues/2009/106/TALK-SOFT
>>
>> 2010/5/8 Seven Du <dujinfang at gmail.com>:
>> > default configuration files of mod_sofia is in conf/sip_profiles
>> >
>> > http://wiki.freeswitch.org/wiki/Sofia
>> > http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files
>> >
>> > 2010/5/8 budi wibowo <bwibowo at gmail.com>:
>> >> thx, after reading the document i still got confusion about  
>> sofia module
>> >> i give example below
>> >>
>> >> Dialing Through A Gateway(SIP Provider)
>> >>
>> >> A gateway is a means for making outbound calls through a SIP  
>> provider. For
>> >> example:
>> >>
>> >> sofia/gateway/mygateway.com/1234
>> >>
>> >> by default i dont find any directory named sofia, where i should  
>> put this
>> >> directory?
>> >> many document i read telling about sofia
>> >> TIA
>> >> budi
>> >>
>> >> On Fri, May 7, 2010 at 12:55 PM, Michael Jerris  
>> <mike at jerris.com> wrote:
>> >>>
>> >>> http://wiki.freeswitch.org/wiki/Dialplan
>> >>> On May 6, 2010, at 11:53 PM, Budi wibowo wrote:
>> >>>
>> >>> Yes I have siphit installed, I tried to make some changes on  
>> dialplan file
>> >>> but call always goes to other server, what should I configure  
>> to implement
>> >>> this outgoing call
>> >>>
>> >>>
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>>
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >
>> >
>> >
>> > --
>> > Blog: http://www.dujinfang.com
>> > Proj:  http://www.freeswitch.org.cn
>> >
>>
>>
>>
>> --
>> Blog: http://www.dujinfang.com
>> Proj:  http://www.freeswitch.org.cn
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From sos at sokhapkin.dyndns.org  Sun May  9 10:12:42 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Sun, 9 May 2010 13:12:42 -0400
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <201004191517.09725.sos@sokhapkin.dyndns.org>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<24412A4A-01E7-4B5D-812E-B879CA65C92F@freeswitch.org>
	<201004191517.09725.sos@sokhapkin.dyndns.org>
Message-ID: <201005091312.42178.sos@sokhapkin.dyndns.org>

Looks like I found a work around. I set <param name="global_heartbeat" 
value="off"/> in nibblebill.conf.xml and execute "nibblebill heartbeat 60" in 
dialplan. No more unbounded memory growth. As soon as I set global_heartbeat, 
memory starts grow again. Go figure...

On Monday 19 April 2010, Sergey Okhapkin wrote:
> It would be the problem if custom_sql_* is set in nibblebill configuration,
> but it's not my case, I do not use custom sql. I'm looking at the code
> more....
> 
> On Monday 19 April 2010, Brian West wrote:
> > No this isn't the problem it would show up on valgrind like crazy.
> >
> > /b
> >
> > On Apr 19, 2010, at 1:25 PM, Sergey Okhapkin wrote:
> > > Yes, I agree that these strdups can be ignored. I believe I found where
> > > the memory problem comes from, the beginning of bill_event function in
> > > mod_nubblebill has SWITCH_STANDARD_STREAM(sql_stream);
> > > But SWITCH_STANDARD_STREAM macro does malloc! I do not see a place in
> > > the function where the memory is freed. Am I right? #define
> > > SWITCH_STANDARD_STREAM(s) memset(&s, 0, sizeof(s)); s.data =
> > > malloc(SWITCH_CMD_CHUNK_LEN); \ switch_assert(s.data);
> > >                                                                       \
> > > memset(s.data, 0, SWITCH_CMD_CHUNK_LEN);
> > >                       \ s.end = s.data;
> > >                                                                       \
> > > s.data_size = SWITCH_CMD_CHUNK_LEN;
> > >                               \ s.write_function =
> > > switch_console_stream_write;                                         \
> > > s.raw_write_function = switch_console_stream_raw_write;
> > >       \ s.alloc_len = SWITCH_CMD_CHUNK_LEN;
> > >                                       \ s.alloc_chunk =
> > > SWITCH_CMD_CHUNK_LEN
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From brian9 at gmail.com  Sat May  8 06:33:02 2010
From: brian9 at gmail.com (Brian Solon)
Date: Sat, 8 May 2010 14:33:02 +0100
Subject: [Freeswitch-users] luarun fs_cli output
Message-ID: <s2p8fa0f3241005080633ue7f5555axf51c8c02cb15da4@mail.gmail.com>

Hi,

This is my first post here so I just wanted to say FreeSWITCH is fantastic.
I'm using it for an art installation which has been up and running for the
last week in a gallery in Sligo, Ireland. ( More info here, for the
curious: http://facebook.com/HelloOperatorIE ).

Just one question for now: how can I see the output of a Lua script when
launched from fs_cli?

-bash-3.2$ pwd
/opt/freeswitch/scripts
-bash-3.2$ lua
Lua 5.1.2  Copyright (C) 1994-2007 Lua.org, PUC-Rio
> require "test"
Hello World
>

I'm probably missing something simple, but when I run it from the CLI all I
get is "+OK":

freeswitch at internal> luarun test.lua
+OK

Thanks,
Brian
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From jan.berger at video24.no  Sat May  8 15:01:58 2010
From: jan.berger at video24.no (Jan Berger)
Date: Sun, 9 May 2010 00:01:58 +0200
Subject: [Freeswitch-users] Alarms-codes
In-Reply-To: <y2zc4d05cbe1005081101ma76fb71do61a7469d1990096c@mail.gmail.com>
References: <017a01caec8a$ee74bb70$cb5e3250$@com>
	<BE3F6D4F-9486-43F3-ACAF-70C84CB001FF@gmail.com>
	<01b101caec94$1f009fd0$5d01df70$@com>
	<w2x87f2f3b91005051355j7c5c775dld43c79c6bf9d61cb@mail.gmail.com>
	<A60A2CFD3B5C41D68FB970C1EB64DFF2@bp1.ad.bp.com>
	<g2n6b65470d1005070808y30d1dce7q1c9b4fec8cc8f058@mail.gmail.com>
	<i2g87f2f3b91005071459i16d8bd2fr7fa57a6304d2c0dc@mail.gmail.com>
	<h2k23f91031005071842t865d2765x595a7a2572f13638@mail.gmail.com>
	<z2o87f2f3b91005072246y39206088g6900a8a81434626c@mail.gmail.com>
	<SNT135-w5FF0C5B6D49D392DF23CDD2F70@phx.gbl>
	<y2zc4d05cbe1005081101ma76fb71do61a7469d1990096c@mail.gmail.com>
Message-ID: <3E6354B851CB4C08BE3592A43F9503A5@dell9400>

You could add an "Alarm" statement in coding, but error's and warning's
serve the same purpose and you would in reality be coding an "alarm" side by
side with these. 

 

Actually - all I really want is the module name + code. Text, severity and
what to do we look up in a config database. 

 

jan

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Moises
Silva
Sent: 8. mai 2010 20:01
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Alarms-codes

 

Hello Jan,

 

I believe you're confusing alarms with logging and those two are not the
same thing and I think should not be mixed. For alarms an special event
should be triggered ( as mentioned in other thread, may be SWITCH_EVENT_TRAP
).

 

mod_freetdm ( FreeSWITCH PSTN module ) is already sending a
SWITCH_EVENT_TRAP when any pstn port becomes alarmed, other modules may
start using it for their own type of alarms. Then we may add the alarm-code
and severity fields. 


Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com



On Sat, May 8, 2010 at 2:47 AM, Jan Berger <janvb at live.com> wrote:

hi,
 
I have playing around with FreeSWITCHf or a while and one of things I am
looking for is unique Alarms-codes
 
FreeSwitch has error, warning, info etc on the logging - but the message
print module name and line-number - it does not incude a proper, unique
alarm-code. Using the combination of module-name and line-number is possible
- but this changes from release to release.
 
The point with alarms is that they are forwarded into
databases/applications/operating senters that monitor telecom equipmenet.
The way this usually is done is that the forwarding mechanism filter on
alarm-code/severity as we don't want to forward junk - but it's a
requirement that we visualise those that affect business. 
 
---
 
It's doable to change logging so that every module insert a unique number -
starting with 1 - this would make modulename + alarm-code an unique
combination that would serve the purpose.
 
Any suggestions?
 
Jan
 
 

  _____  

Hotmail: Trusted email with Microsoft's powerful SPAM protection. Sign up
<https://signup.live.com/signup.aspx?id=60969>  now.


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

 

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From tony.tin at noahmedia.com.hk  Sun May  9 06:16:23 2010
From: tony.tin at noahmedia.com.hk (Tony Tin)
Date: Sun, 9 May 2010 21:16:23 +0800
Subject: [Freeswitch-users] short ringback tone when answer
Message-ID: <j2g191e3b11005090616ndd0b0ef8g37f60c1f1016ab4f@mail.gmail.com>

Hi All,

I've just setup a new IVRS with FreeSwitch and a Digium TE220 card (which is
connected to a 4ess T1). Every thing works fine so far except one little
problem that annoys me. Every time I dial in the ivrs, there is a very short
(0.5 second) ringback tone exactly while the "<action
application="answer"/>" line is executed in the dial plan. I want to disable
this ringback tone because the ivrs will answer the call immediately, the
ringback tone is really unnecessary in this case, but I fail to do that. I
try to set the variable "ringback" to change the ringback tone, it's also
not working. Could anyone please help.

I'm using the native pri stack.
The default.xml containts only below lines, and attached is the log file.

       <extension name="testing">
          <condition field="destination_number" expression="^(6.*)$">
                <action application="sleep" data="3000"/>
                <action application="answer"/>
                <action application="sleep" data="3000"/>
          </condition>
       </extension>

Regards,
Tony
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From jan.berger at video24.no  Sun May  9 10:30:04 2010
From: jan.berger at video24.no (Jan Berger)
Date: Sun, 9 May 2010 19:30:04 +0200
Subject: [Freeswitch-users] Compile errors on VC 2008 express
Message-ID: <63CD9C6BFA234C90A6E0B185CCBCE806@dell9400>

Hi,

 

I have been working in 1.0.5 that compiles fine, but as I downloaded the
latest version from Git I get compile errors in libsofia and others. Are
others experiencing this, or is it just me?

 

Jan

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From red.rain.seven at gmail.com  Sun May  9 11:01:08 2010
From: red.rain.seven at gmail.com (Henry Huang)
Date: Mon, 10 May 2010 02:01:08 +0800
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <201005091312.42178.sos@sokhapkin.dyndns.org>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<24412A4A-01E7-4B5D-812E-B879CA65C92F@freeswitch.org>
	<201004191517.09725.sos@sokhapkin.dyndns.org>
	<201005091312.42178.sos@sokhapkin.dyndns.org>
Message-ID: <p2j59ad9ca11005091101vd9370816p1109fbf2fc323d9f@mail.gmail.com>

Sergey:

Which version of nibbliebill are you using. We are having memory leak issue
as well with freeswitch 1.0.4 , but I haven't try to disable nibblebill in
our test, I will do that now. But since we are already doing what you say
about "<param name="global_heartbeat"
value="off"/> in nibblebill.conf.xml and execute "nibblebill heartbeat 60"
in
dialplan."  in our current environment, and still getting memory leak, I am
wondering if you are using nibblebill that comes with the freeswitch 1.0.6
version.

Thanks,



On Mon, May 10, 2010 at 1:12 AM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> Looks like I found a work around. I set <param name="global_heartbeat"
> value="off"/> in nibblebill.conf.xml and execute "nibblebill heartbeat 60"
> in
> dialplan. No more unbounded memory growth. As soon as I set
> global_heartbeat,
> memory starts grow again. Go figure...
>
> On Monday 19 April 2010, Sergey Okhapkin wrote:
> > It would be the problem if custom_sql_* is set in nibblebill
> configuration,
> > but it's not my case, I do not use custom sql. I'm looking at the code
> > more....
> >
> > On Monday 19 April 2010, Brian West wrote:
> > > No this isn't the problem it would show up on valgrind like crazy.
> > >
> > > /b
> > >
> > > On Apr 19, 2010, at 1:25 PM, Sergey Okhapkin wrote:
> > > > Yes, I agree that these strdups can be ignored. I believe I found
> where
> > > > the memory problem comes from, the beginning of bill_event function
> in
> > > > mod_nubblebill has SWITCH_STANDARD_STREAM(sql_stream);
> > > > But SWITCH_STANDARD_STREAM macro does malloc! I do not see a place in
> > > > the function where the memory is freed. Am I right? #define
> > > > SWITCH_STANDARD_STREAM(s) memset(&s, 0, sizeof(s)); s.data =
> > > > malloc(SWITCH_CMD_CHUNK_LEN); \ switch_assert(s.data);
> > > >
> \
> > > > memset(s.data, 0, SWITCH_CMD_CHUNK_LEN);
> > > >                       \ s.end = s.data;
> > > >
> \
> > > > s.data_size = SWITCH_CMD_CHUNK_LEN;
> > > >                               \ s.write_function =
> > > > switch_console_stream_write;
> \
> > > > s.raw_write_function = switch_console_stream_raw_write;
> > > >       \ s.alloc_len = SWITCH_CMD_CHUNK_LEN;
> > > >                                       \ s.alloc_chunk =
> > > > SWITCH_CMD_CHUNK_LEN
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Henry Huang
VoIP & Open Source software Consultant
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From david.ponzone at gmail.com  Sun May  9 11:05:53 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Sun, 9 May 2010 20:05:53 +0200
Subject: [Freeswitch-users] luarun fs_cli output
In-Reply-To: <s2p8fa0f3241005080633ue7f5555axf51c8c02cb15da4@mail.gmail.com>
References: <s2p8fa0f3241005080633ue7f5555axf51c8c02cb15da4@mail.gmail.com>
Message-ID: <8F5F793A-902D-4932-B17A-7E172E039976@gmail.com>

AFAIK, You can't.
You may send the required output to FS console with the required API  
call (consoleLog).

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 08/05/2010 ? 15:33, Brian Solon a ?crit :

> Hi,
>
> This is my first post here so I just wanted to say FreeSWITCH is  
> fantastic. I'm using it for an art installation which has been up  
> and running for the last week in a gallery in Sligo, Ireland. ( More  
> info here, for the curious: http://facebook.com/HelloOperatorIE ).
>
> Just one question for now: how can I see the output of a Lua script  
> when launched from fs_cli?
>
> -bash-3.2$ pwd
> /opt/freeswitch/scripts
> -bash-3.2$ lua
> Lua 5.1.2  Copyright (C) 1994-2007 Lua.org, PUC-Rio
> > require "test"
> Hello World
> >
>
> I'm probably missing something simple, but when I run it from the  
> CLI all I get is "+OK":
>
> freeswitch at internal> luarun test.lua
> +OK
>
> Thanks,
> Brian
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From sos at sokhapkin.dyndns.org  Sun May  9 11:11:53 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Sun, 9 May 2010 14:11:53 -0400
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <p2j59ad9ca11005091101vd9370816p1109fbf2fc323d9f@mail.gmail.com>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<201005091312.42178.sos@sokhapkin.dyndns.org>
	<p2j59ad9ca11005091101vd9370816p1109fbf2fc323d9f@mail.gmail.com>
Message-ID: <201005091411.53990.sos@sokhapkin.dyndns.org>

I run git trunk. I afraid the workaround doesn't work, memory usage still 
grows, but slower than with global_heartbeat set.

On Sunday 09 May 2010, Henry Huang wrote:
> Sergey:
> 
> Which version of nibbliebill are you using. We are having memory leak issue
> as well with freeswitch 1.0.4 , but I haven't try to disable nibblebill in
> our test, I will do that now. But since we are already doing what you say
> about "<param name="global_heartbeat"
> value="off"/> in nibblebill.conf.xml and execute "nibblebill heartbeat 60"
> in
> dialplan."  in our current environment, and still getting memory leak, I am
> wondering if you are using nibblebill that comes with the freeswitch 1.0.6
> version.
> 
> Thanks,
> 
> 
> 
> On Mon, May 10, 2010 at 1:12 AM, Sergey Okhapkin
> 
> <sos at sokhapkin.dyndns.org>wrote:
> > Looks like I found a work around. I set <param name="global_heartbeat"
> > value="off"/> in nibblebill.conf.xml and execute "nibblebill heartbeat
> > 60" in
> > dialplan. No more unbounded memory growth. As soon as I set
> > global_heartbeat,
> > memory starts grow again. Go figure...
> >
> > On Monday 19 April 2010, Sergey Okhapkin wrote:
> > > It would be the problem if custom_sql_* is set in nibblebill
> >
> > configuration,
> >
> > > but it's not my case, I do not use custom sql. I'm looking at the code
> > > more....
> > >
> > > On Monday 19 April 2010, Brian West wrote:
> > > > No this isn't the problem it would show up on valgrind like crazy.
> > > >
> > > > /b
> > > >
> > > > On Apr 19, 2010, at 1:25 PM, Sergey Okhapkin wrote:
> > > > > Yes, I agree that these strdups can be ignored. I believe I found
> >
> > where
> >
> > > > > the memory problem comes from, the beginning of bill_event function
> >
> > in
> >
> > > > > mod_nubblebill has SWITCH_STANDARD_STREAM(sql_stream);
> > > > > But SWITCH_STANDARD_STREAM macro does malloc! I do not see a place
> > > > > in the function where the memory is freed. Am I right? #define
> > > > > SWITCH_STANDARD_STREAM(s) memset(&s, 0, sizeof(s)); s.data =
> > > > > malloc(SWITCH_CMD_CHUNK_LEN); \ switch_assert(s.data);
> >
> > \
> >
> > > > > memset(s.data, 0, SWITCH_CMD_CHUNK_LEN);
> > > > >                       \ s.end = s.data;
> >
> > \
> >
> > > > > s.data_size = SWITCH_CMD_CHUNK_LEN;
> > > > >                               \ s.write_function =
> > > > > switch_console_stream_write;
> >
> > \
> >
> > > > > s.raw_write_function = switch_console_stream_raw_write;
> > > > >       \ s.alloc_len = SWITCH_CMD_CHUNK_LEN;
> > > > >                                       \ s.alloc_chunk =
> > > > > SWITCH_CMD_CHUNK_LEN
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
> > >s http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 



From peter.olsson at visionutveckling.se  Sun May  9 11:29:28 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sun, 9 May 2010 20:29:28 +0200
Subject: [Freeswitch-users] Compile errors on VC 2008 express
In-Reply-To: <63CD9C6BFA234C90A6E0B185CCBCE806@dell9400>
References: <63CD9C6BFA234C90A6E0B185CCBCE806@dell9400>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5576CE4DC6@cooper>

This has been discussed on the list for a couple of times, but read the instructions on http://wiki.freeswitch.org/wiki/Git_Install#Windows.

The basic idea is to set autocrlf to false, so LF won't be converted in a bad way on Windows.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Jan Berger [jan.berger at video24.no]
Skickat: den 9 maj 2010 19:30
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] Compile errors on VC 2008 express

Hi,

I have been working in 1.0.5 that compiles fine, but as I downloaded the latest version from Git I get compile errors in libsofia and others. Are others experiencing this, or is it just me?

Jan
!DSPAM:4be6f3b832932140028801!


From brian9 at gmail.com  Sun May  9 11:32:16 2010
From: brian9 at gmail.com (Brian Solon)
Date: Sun, 9 May 2010 19:32:16 +0100
Subject: [Freeswitch-users] luarun fs_cli output
In-Reply-To: <8F5F793A-902D-4932-B17A-7E172E039976@gmail.com>
References: <s2p8fa0f3241005080633ue7f5555axf51c8c02cb15da4@mail.gmail.com> 
	<8F5F793A-902D-4932-B17A-7E172E039976@gmail.com>
Message-ID: <z2o8fa0f3241005091132h7b970f91n8c718fa34bf5cec@mail.gmail.com>

Thanks, I'll try that.

On 9 May 2010 19:05, David Ponzone <david.ponzone at gmail.com> wrote:

> AFAIK, You can't.
> You may send the required output to FS console with the required API call
> (consoleLog).
>
>  David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 08/05/2010 ? 15:33, Brian Solon a ?crit :
>
> Hi,
>
> This is my first post here so I just wanted to say FreeSWITCH is fantastic.
> I'm using it for an art installation which has been up and running for the
> last week in a gallery in Sligo, Ireland. ( More info here, for the
> curious: http://facebook.com/HelloOperatorIE ).
>
> Just one question for now: how can I see the output of a Lua script when
> launched from fs_cli?
>
> -bash-3.2$ pwd
> /opt/freeswitch/scripts
> -bash-3.2$ lua
> Lua 5.1.2  Copyright (C) 1994-2007 Lua.org, PUC-Rio
> > require "test"
> Hello World
> >
>
> I'm probably missing something simple, but when I run it from the CLI all I
> get is "+OK":
>
> freeswitch at internal> luarun test.lua
> +OK
>
> Thanks,
> Brian
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From peter.olsson at visionutveckling.se  Sun May  9 11:33:42 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sun, 9 May 2010 20:33:42 +0200
Subject: [Freeswitch-users] short ringback tone when answer
In-Reply-To: <j2g191e3b11005090616ndd0b0ef8g37f60c1f1016ab4f@mail.gmail.com>
References: <j2g191e3b11005090616ndd0b0ef8g37f60c1f1016ab4f@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5576CE4DC7@cooper>

The equipment connected in the other en of the T1 is probably playing these tones for you, 500ms is probably the time it takes for the called to be connected successfully.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Tony Tin [tony.tin at noahmedia.com.hk]
Skickat: den 9 maj 2010 15:16
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] short ringback tone when answer

Hi All,

I've just setup a new IVRS with FreeSwitch and a Digium TE220 card (which is connected to a 4ess T1). Every thing works fine so far except one little problem that annoys me. Every time I dial in the ivrs, there is a very short (0.5 second) ringback tone exactly while the "<action application="answer"/>" line is executed in the dial plan. I want to disable this ringback tone because the ivrs will answer the call immediately, the ringback tone is really unnecessary in this case, but I fail to do that. I try to set the variable "ringback" to change the ringback tone, it's also not working. Could anyone please help.

I'm using the native pri stack.
The default.xml containts only below lines, and attached is the log file.

       <extension name="testing">
          <condition field="destination_number" expression="^(6.*)$">
                <action application="sleep" data="3000"/>
                <action application="answer"/>
                <action application="sleep" data="3000"/>
          </condition>
       </extension>

Regards,
Tony

!DSPAM:4be6f37432931620317181!


From brian at freeswitch.org  Sun May  9 11:57:34 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 9 May 2010 13:57:34 -0500
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <201005091411.53990.sos@sokhapkin.dyndns.org>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<201005091312.42178.sos@sokhapkin.dyndns.org>
	<p2j59ad9ca11005091101vd9370816p1109fbf2fc323d9f@mail.gmail.com>
	<201005091411.53990.sos@sokhapkin.dyndns.org>
Message-ID: <3121F983-02B1-4236-B143-CCA74D3D8BF9@freeswitch.org>

Please use valgrind and see where its leaking then open a jira.

Nobody has been able to reproduce this in a lab nor provide any details to assist in finding the issue... All I have seen is people complaining about it and not doing what they should debugging the issue and reporting it.

/b

On May 9, 2010, at 1:11 PM, Sergey Okhapkin wrote:

> I run git trunk. I afraid the workaround doesn't work, memory usage still 
> grows, but slower than with global_heartbeat set.



From sos at sokhapkin.dyndns.org  Sun May  9 12:10:46 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Sun, 9 May 2010 15:10:46 -0400
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <3121F983-02B1-4236-B143-CCA74D3D8BF9@freeswitch.org>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<201005091411.53990.sos@sokhapkin.dyndns.org>
	<3121F983-02B1-4236-B143-CCA74D3D8BF9@freeswitch.org>
Message-ID: <201005091510.46511.sos@sokhapkin.dyndns.org>

Valgrind output shows no significant leaks. This was discussed already in this 
thread a month ago.

On Sunday 09 May 2010, Brian West wrote:
> Please use valgrind and see where its leaking then open a jira.
> 
> Nobody has been able to reproduce this in a lab nor provide any details to
>  assist in finding the issue... All I have seen is people complaining about
>  it and not doing what they should debugging the issue and reporting it.
> 
> /b
> 
> On May 9, 2010, at 1:11 PM, Sergey Okhapkin wrote:
> > I run git trunk. I afraid the workaround doesn't work, memory usage still
> > grows, but slower than with global_heartbeat set.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From jbrucehopkins at gmail.com  Sun May  9 12:19:20 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Sun, 9 May 2010 20:19:20 +0100
Subject: [Freeswitch-users] Anyone in UK want a free UTStarcom F1000G wifi
	phone?
Message-ID: <AANLkTikXhMIE8ZYXnxzO6faAXcCAt6aTYrBDbbRbqMw_@mail.gmail.com>

It's as new, in its original packaging - but I'm not going to need it and
it's too obscure to sell through ebay.

So if you want it, it's yours.

I'm moving house and need to cut down on the amount of unused stuff here -
so I'll even pay the UK postage to avoid throwing it away.

Any taker? Seriously !

Best regards
Bruce
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From stevendt at primrosebank.net  Sun May  9 12:31:42 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Sun, 9 May 2010 20:31:42 +0100
Subject: [Freeswitch-users] Anyone in UK want a free UTStarcom F1000G
	wifiphone?
References: <AANLkTikXhMIE8ZYXnxzO6faAXcCAt6aTYrBDbbRbqMw_@mail.gmail.com>
Message-ID: <87E1EE0F617A4C7482DCB5ABEE19E8CA@bp1.ad.bp.com>

Wow - if this is a serious offer - then I'm your man ! 

Please please please ?

how do we get in touch ?

regards
Dave


  ----- Original Message ----- 
  From: Bruce Hopkins 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Sunday, May 09, 2010 8:19 PM
  Subject: [Freeswitch-users] Anyone in UK want a free UTStarcom F1000G wifiphone?


  It's as new, in its original packaging - but I'm not going to need it and it's too obscure to sell through ebay.

  So if you want it, it's yours.

  I'm moving house and need to cut down on the amount of unused stuff here - so I'll even pay the UK postage to avoid throwing it away.

  Any taker? Seriously !

  Best regards
  Bruce


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


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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From brian at freeswitch.org  Sun May  9 12:37:24 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 9 May 2010 14:37:24 -0500
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <201005091510.46511.sos@sokhapkin.dyndns.org>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<201005091411.53990.sos@sokhapkin.dyndns.org>
	<3121F983-02B1-4236-B143-CCA74D3D8BF9@freeswitch.org>
	<201005091510.46511.sos@sokhapkin.dyndns.org>
Message-ID: <2267B338-D811-43B0-AC98-8DDC7A8480A8@freeswitch.org>

What are you calling significant memory usage?

/b

On May 9, 2010, at 2:10 PM, Sergey Okhapkin wrote:

> Valgrind output shows no significant leaks. This was discussed already in this 
> thread a month ago.
> 
> On Sunday 09 May 2010, Brian West wrote:
>> Please use valgrind and see where its leaking then open a jira.
>> 
>> Nobody has been able to reproduce this in a lab nor provide any details to
>> assist in finding the issue... All I have seen is people complaining about
>> it and not doing what they should debugging the issue and reporting it.
>> 
>> /b



From jbrucehopkins at gmail.com  Sun May  9 12:44:08 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Sun, 9 May 2010 20:44:08 +0100
Subject: [Freeswitch-users] Anyone in UK want a free UTStarcom F1000G
	wifiphone?
In-Reply-To: <87E1EE0F617A4C7482DCB5ABEE19E8CA@bp1.ad.bp.com>
References: <AANLkTikXhMIE8ZYXnxzO6faAXcCAt6aTYrBDbbRbqMw_@mail.gmail.com>
	<87E1EE0F617A4C7482DCB5ABEE19E8CA@bp1.ad.bp.com>
Message-ID: <AANLkTikUqB-JP8Pp9zVisC14W24ooYkExIhHYbeFMf4a@mail.gmail.com>

Dave - you were beaten to it by 1 minute I'm afraid.

Sorry
Bruce

On 9 May 2010 20:31, Dave Stevenson <stevendt at primrosebank.net> wrote:

>  Wow - if this is a serious offer - then I'm your man !
>
> Please please please ?
>
> how do we get in touch ?
>
> regards
> Dave
>
>
>
> ----- Original Message -----
> *From:* Bruce Hopkins <jbrucehopkins at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Sunday, May 09, 2010 8:19 PM
> *Subject:* [Freeswitch-users] Anyone in UK want a free UTStarcom F1000G
> wifiphone?
>
> It's as new, in its original packaging - but I'm not going to need it and
> it's too obscure to sell through ebay.
>
> So if you want it, it's yours.
>
> I'm moving house and need to cut down on the amount of unused stuff here -
> so I'll even pay the UK postage to avoid throwing it away.
>
> Any taker? Seriously !
>
> Best regards
> Bruce
>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From stevendt at primrosebank.net  Sun May  9 12:48:42 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Sun, 9 May 2010 20:48:42 +0100
Subject: [Freeswitch-users] Anyone in UK want a free UTStarcom
	F1000Gwifiphone?
References: <AANLkTikXhMIE8ZYXnxzO6faAXcCAt6aTYrBDbbRbqMw_@mail.gmail.com><87E1EE0F617A4C7482DCB5ABEE19E8CA@bp1.ad.bp.com>
	<AANLkTikUqB-JP8Pp9zVisC14W24ooYkExIhHYbeFMf4a@mail.gmail.com>
Message-ID: <7A0AFD42891D42CA8D1C7F2B9A75731C@bp1.ad.bp.com>

OK, thanks anyway,

I thought it was too good to be true :-(

regards
Dave
  ----- Original Message ----- 
  From: Bruce Hopkins 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Sunday, May 09, 2010 8:44 PM
  Subject: Re: [Freeswitch-users] Anyone in UK want a free UTStarcom F1000Gwifiphone?


  Dave - you were beaten to it by 1 minute I'm afraid.

  Sorry
  Bruce


  On 9 May 2010 20:31, Dave Stevenson <stevendt at primrosebank.net> wrote:

    Wow - if this is a serious offer - then I'm your man ! 

    Please please please ?

    how do we get in touch ?

    regards
    Dave


      ----- Original Message ----- 
      From: Bruce Hopkins 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Sunday, May 09, 2010 8:19 PM
      Subject: [Freeswitch-users] Anyone in UK want a free UTStarcom F1000G wifiphone?


      It's as new, in its original packaging - but I'm not going to need it and it's too obscure to sell through ebay.

      So if you want it, it's yours.

      I'm moving house and need to cut down on the amount of unused stuff here - so I'll even pay the UK postage to avoid throwing it away.

      Any taker? Seriously !

      Best regards
      Bruce



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


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


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


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From sos at sokhapkin.dyndns.org  Sun May  9 12:49:40 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Sun, 9 May 2010 15:49:40 -0400
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <2267B338-D811-43B0-AC98-8DDC7A8480A8@freeswitch.org>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<201005091510.46511.sos@sokhapkin.dyndns.org>
	<2267B338-D811-43B0-AC98-8DDC7A8480A8@freeswitch.org>
Message-ID: <201005091549.40332.sos@sokhapkin.dyndns.org>

RSS grows to 300M on 100 concurrent calls during several hours. With 
mod_nibblebill disabled RSS stays at about 60-70M

On Sunday 09 May 2010, Brian West wrote:
> What are you calling significant memory usage?
> 
> /b
> 
> On May 9, 2010, at 2:10 PM, Sergey Okhapkin wrote:
> > Valgrind output shows no significant leaks. This was discussed already in
> > this thread a month ago.
> >
> > On Sunday 09 May 2010, Brian West wrote:
> >> Please use valgrind and see where its leaking then open a jira.
> >>
> >> Nobody has been able to reproduce this in a lab nor provide any details
> >> to assist in finding the issue... All I have seen is people complaining
> >> about it and not doing what they should debugging the issue and
> >> reporting it.
> >>
> >> /b
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From jbrucehopkins at gmail.com  Sun May  9 12:57:45 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Sun, 9 May 2010 20:57:45 +0100
Subject: [Freeswitch-users] Anyone in UK want a free VXI Passport 20 V DC
	headset?
Message-ID: <AANLkTin0-YZK9bTkSg83Jhmf1KJrll8vqJTbB3BkhUqm@mail.gmail.com>

Ok, flushed with the success of my wifi phone giveaway ...

I also have a pretty much as-new binaural VXI Passport headset using up too
much space in the cupboard.

It comes with every different bottom-cable connector I could think of when I
was doing some testing with it.

There are three different flavours of RJ9, plus 3.2mm, plus 2.5mm - so
should be able to connect to about any phone.

If you are inthe UK and can offer it and the cables a good home, it's yours.

Any takers?

Bruce
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From jbrucehopkins at gmail.com  Sun May  9 12:58:36 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Sun, 9 May 2010 20:58:36 +0100
Subject: [Freeswitch-users] Anyone in UK want a free UTStarcom
	F1000Gwifiphone?
In-Reply-To: <7A0AFD42891D42CA8D1C7F2B9A75731C@bp1.ad.bp.com>
References: <AANLkTikXhMIE8ZYXnxzO6faAXcCAt6aTYrBDbbRbqMw_@mail.gmail.com>
	<87E1EE0F617A4C7482DCB5ABEE19E8CA@bp1.ad.bp.com>
	<AANLkTikUqB-JP8Pp9zVisC14W24ooYkExIhHYbeFMf4a@mail.gmail.com>
	<7A0AFD42891D42CA8D1C7F2B9A75731C@bp1.ad.bp.com>
Message-ID: <AANLkTin5V4EQEDbru3QaqJ3n69cK5nETdZxbybZAp8fW@mail.gmail.com>

Sorry mate.   It was true - just fleetingly so !

On 9 May 2010 20:48, Dave Stevenson <stevendt at primrosebank.net> wrote:

>  OK, thanks anyway,
>
> I thought it was too good to be true :-(
>
> regards
> Dave
>
> ----- Original Message -----
> *From:* Bruce Hopkins <jbrucehopkins at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Sunday, May 09, 2010 8:44 PM
> *Subject:* Re: [Freeswitch-users] Anyone in UK want a free UTStarcom
> F1000Gwifiphone?
>
> Dave - you were beaten to it by 1 minute I'm afraid.
>
> Sorry
> Bruce
>
> On 9 May 2010 20:31, Dave Stevenson <stevendt at primrosebank.net> wrote:
>
>>  Wow - if this is a serious offer - then I'm your man !
>>
>> Please please please ?
>>
>> how do we get in touch ?
>>
>> regards
>> Dave
>>
>>
>>
>>   ----- Original Message -----
>> *From:* Bruce Hopkins <jbrucehopkins at gmail.com>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Sent:* Sunday, May 09, 2010 8:19 PM
>> *Subject:* [Freeswitch-users] Anyone in UK want a free UTStarcom F1000G
>> wifiphone?
>>
>> It's as new, in its original packaging - but I'm not going to need it and
>> it's too obscure to sell through ebay.
>>
>> So if you want it, it's yours.
>>
>> I'm moving house and need to cut down on the amount of unused stuff here -
>> so I'll even pay the UK postage to avoid throwing it away.
>>
>> Any taker? Seriously !
>>
>> Best regards
>> Bruce
>>
>> ------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From stevendt at primrosebank.net  Sun May  9 13:13:55 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Sun, 9 May 2010 21:13:55 +0100
Subject: [Freeswitch-users] Anyone in UK want a free VXI Passport 20 V
	DCheadset?
References: <AANLkTin0-YZK9bTkSg83Jhmf1KJrll8vqJTbB3BkhUqm@mail.gmail.com>
Message-ID: <8CC040E5AC74402AB6840F55F5195816@bp1.ad.bp.com>

OK - I'm all ears !

(well, I';ve only one, but still !)

regards
Dave
  ----- Original Message ----- 
  From: Bruce Hopkins 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Sunday, May 09, 2010 8:57 PM
  Subject: [Freeswitch-users] Anyone in UK want a free VXI Passport 20 V DCheadset?


  Ok, flushed with the success of my wifi phone giveaway ...

  I also have a pretty much as-new binaural VXI Passport headset using up too much space in the cupboard.

  It comes with every different bottom-cable connector I could think of when I was doing some testing with it.

  There are three different flavours of RJ9, plus 3.2mm, plus 2.5mm - so should be able to connect to about any phone.

  If you are inthe UK and can offer it and the cables a good home, it's yours.

  Any takers?

  Bruce




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


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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From stevendt at primrosebank.net  Sun May  9 13:16:37 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Sun, 9 May 2010 21:16:37 +0100
Subject: [Freeswitch-users] Anyone in UK want a free
	UTStarcomF1000Gwifiphone?
References: <AANLkTikXhMIE8ZYXnxzO6faAXcCAt6aTYrBDbbRbqMw_@mail.gmail.com><87E1EE0F617A4C7482DCB5ABEE19E8CA@bp1.ad.bp.com><AANLkTikUqB-JP8Pp9zVisC14W24ooYkExIhHYbeFMf4a@mail.gmail.com><7A0AFD42891D42CA8D1C7F2B9A75731C@bp1.ad.bp.com>
	<AANLkTin5V4EQEDbru3QaqJ3n69cK5nETdZxbybZAp8fW@mail.gmail.com>
Message-ID: <551DE642D9DD49D2BF55F44997822BB6@bp1.ad.bp.com>

Yeah - I was just too slow !

I knew I shouldn't have gone and made that coffee :-)

  ----- Original Message ----- 
  From: Bruce Hopkins 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Sunday, May 09, 2010 8:58 PM
  Subject: Re: [Freeswitch-users] Anyone in UK want a free UTStarcomF1000Gwifiphone?


  Sorry mate.   It was true - just fleetingly so !


  On 9 May 2010 20:48, Dave Stevenson <stevendt at primrosebank.net> wrote:

    OK, thanks anyway,

    I thought it was too good to be true :-(

    regards
    Dave
      ----- Original Message ----- 
      From: Bruce Hopkins 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Sunday, May 09, 2010 8:44 PM
      Subject: Re: [Freeswitch-users] Anyone in UK want a free UTStarcom F1000Gwifiphone?


      Dave - you were beaten to it by 1 minute I'm afraid.

      Sorry
      Bruce


      On 9 May 2010 20:31, Dave Stevenson <stevendt at primrosebank.net> wrote:

        Wow - if this is a serious offer - then I'm your man ! 

        Please please please ?

        how do we get in touch ?

        regards
        Dave


          ----- Original Message ----- 
          From: Bruce Hopkins 
          To: freeswitch-users at lists.freeswitch.org 
          Sent: Sunday, May 09, 2010 8:19 PM
          Subject: [Freeswitch-users] Anyone in UK want a free UTStarcom F1000G wifiphone?


          It's as new, in its original packaging - but I'm not going to need it and it's too obscure to sell through ebay.

          So if you want it, it's yours.

          I'm moving house and need to cut down on the amount of unused stuff here - so I'll even pay the UK postage to avoid throwing it away.

          Any taker? Seriously !

          Best regards
          Bruce



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


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


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        FreeSWITCH-users at lists.freeswitch.org
        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From jbrucehopkins at gmail.com  Sun May  9 13:30:21 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Sun, 9 May 2010 21:30:21 +0100
Subject: [Freeswitch-users] Anyone in UK want a free VXI Passport 20 V
	DCheadset?
In-Reply-To: <8CC040E5AC74402AB6840F55F5195816@bp1.ad.bp.com>
References: <AANLkTin0-YZK9bTkSg83Jhmf1KJrll8vqJTbB3BkhUqm@mail.gmail.com>
	<8CC040E5AC74402AB6840F55F5195816@bp1.ad.bp.com>
Message-ID: <AANLkTinJfCFGSeha_e1XZcoBLrgG4ent_P4fTwh_zkj1@mail.gmail.com>

It's yours Dave.

Let me know the address and I'll post it tomorrow.

cheers
Bruce

On 9 May 2010 21:13, Dave Stevenson <stevendt at primrosebank.net> wrote:

>  OK - I'm all ears !
>
> (well, I';ve only one, but still !)
>
> regards
> Dave
>
> ----- Original Message -----
> *From:* Bruce Hopkins <jbrucehopkins at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Sunday, May 09, 2010 8:57 PM
> *Subject:* [Freeswitch-users] Anyone in UK want a free VXI Passport 20 V
> DCheadset?
>
> Ok, flushed with the success of my wifi phone giveaway ...
>
> I also have a pretty much as-new binaural VXI Passport headset using up too
> much space in the cupboard.
>
> It comes with every different bottom-cable connector I could think of when
> I was doing some testing with it.
>
> There are three different flavours of RJ9, plus 3.2mm, plus 2.5mm - so
> should be able to connect to about any phone.
>
> If you are inthe UK and can offer it and the cables a good home, it's
> yours.
>
> Any takers?
>
> Bruce
>
>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jbrucehopkins at gmail.com  Sun May  9 13:32:28 2010
From: jbrucehopkins at gmail.com (Bruce Hopkins)
Date: Sun, 9 May 2010 21:32:28 +0100
Subject: [Freeswitch-users] Anyone in UK want a free
	UTStarcomF1000Gwifiphone?
In-Reply-To: <551DE642D9DD49D2BF55F44997822BB6@bp1.ad.bp.com>
References: <AANLkTikXhMIE8ZYXnxzO6faAXcCAt6aTYrBDbbRbqMw_@mail.gmail.com>
	<87E1EE0F617A4C7482DCB5ABEE19E8CA@bp1.ad.bp.com>
	<AANLkTikUqB-JP8Pp9zVisC14W24ooYkExIhHYbeFMf4a@mail.gmail.com>
	<7A0AFD42891D42CA8D1C7F2B9A75731C@bp1.ad.bp.com>
	<AANLkTin5V4EQEDbru3QaqJ3n69cK5nETdZxbybZAp8fW@mail.gmail.com>
	<551DE642D9DD49D2BF55F44997822BB6@bp1.ad.bp.com>
Message-ID: <AANLkTinnpxpxU_sgo4B-rTmmsHU-Y7r_KKAvf6vf4-3a@mail.gmail.com>

I guess it's a case of he who caffeinates is lost ;-)

On 9 May 2010 21:16, Dave Stevenson <stevendt at primrosebank.net> wrote:

>  Yeah - I was just too slow !
>
> I knew I shouldn't have gone and made that coffee :-)
>
>
> ----- Original Message -----
> *From:* Bruce Hopkins <jbrucehopkins at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Sunday, May 09, 2010 8:58 PM
> *Subject:* Re: [Freeswitch-users] Anyone in UK want a free
> UTStarcomF1000Gwifiphone?
>
> Sorry mate.   It was true - just fleetingly so !
>
> On 9 May 2010 20:48, Dave Stevenson <stevendt at primrosebank.net> wrote:
>
>>  OK, thanks anyway,
>>
>> I thought it was too good to be true :-(
>>
>> regards
>> Dave
>>
>>  ----- Original Message -----
>> *From:* Bruce Hopkins <jbrucehopkins at gmail.com>
>> *To:* freeswitch-users at lists.freeswitch.org
>>   *Sent:* Sunday, May 09, 2010 8:44 PM
>> *Subject:* Re: [Freeswitch-users] Anyone in UK want a free UTStarcom
>> F1000Gwifiphone?
>>
>> Dave - you were beaten to it by 1 minute I'm afraid.
>>
>> Sorry
>> Bruce
>>
>> On 9 May 2010 20:31, Dave Stevenson <stevendt at primrosebank.net> wrote:
>>
>>>  Wow - if this is a serious offer - then I'm your man !
>>>
>>> Please please please ?
>>>
>>> how do we get in touch ?
>>>
>>> regards
>>> Dave
>>>
>>>
>>>
>>>   ----- Original Message -----
>>> *From:* Bruce Hopkins <jbrucehopkins at gmail.com>
>>> *To:* freeswitch-users at lists.freeswitch.org
>>> *Sent:* Sunday, May 09, 2010 8:19 PM
>>> *Subject:* [Freeswitch-users] Anyone in UK want a free UTStarcom F1000G
>>> wifiphone?
>>>
>>> It's as new, in its original packaging - but I'm not going to need it and
>>> it's too obscure to sell through ebay.
>>>
>>> So if you want it, it's yours.
>>>
>>> I'm moving house and need to cut down on the amount of unused stuff here
>>> - so I'll even pay the UK postage to avoid throwing it away.
>>>
>>> Any taker? Seriously !
>>>
>>> Best regards
>>> Bruce
>>>
>>> ------------------------------
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>  ------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From stevendt at primrosebank.net  Sun May  9 13:34:46 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Sun, 9 May 2010 21:34:46 +0100
Subject: [Freeswitch-users] Anyone in UK want a free VXI Passport 20
	VDCheadset?
References: <AANLkTin0-YZK9bTkSg83Jhmf1KJrll8vqJTbB3BkhUqm@mail.gmail.com><8CC040E5AC74402AB6840F55F5195816@bp1.ad.bp.com>
	<AANLkTinJfCFGSeha_e1XZcoBLrgG4ent_P4fTwh_zkj1@mail.gmail.com>
Message-ID: <133AD3F17F444D29993AAB1287AD8A9F@bp1.ad.bp.com>

Whee !

thanks Bruce - address to follow in PM,

regards
Dave
  ----- Original Message ----- 
  From: Bruce Hopkins 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Sunday, May 09, 2010 9:30 PM
  Subject: Re: [Freeswitch-users] Anyone in UK want a free VXI Passport 20 VDCheadset?


  It's yours Dave.

  Let me know the address and I'll post it tomorrow.

  cheers
  Bruce


  On 9 May 2010 21:13, Dave Stevenson <stevendt at primrosebank.net> wrote:

    OK - I'm all ears !

    (well, I';ve only one, but still !)

    regards
    Dave
      ----- Original Message ----- 
      From: Bruce Hopkins 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Sunday, May 09, 2010 8:57 PM
      Subject: [Freeswitch-users] Anyone in UK want a free VXI Passport 20 V DCheadset?


      Ok, flushed with the success of my wifi phone giveaway ...

      I also have a pretty much as-new binaural VXI Passport headset using up too much space in the cupboard.

      It comes with every different bottom-cable connector I could think of when I was doing some testing with it.

      There are three different flavours of RJ9, plus 3.2mm, plus 2.5mm - so should be able to connect to about any phone.

      If you are inthe UK and can offer it and the cables a good home, it's yours.

      Any takers?

      Bruce





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From stevendt at primrosebank.net  Sun May  9 13:37:51 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Sun, 9 May 2010 21:37:51 +0100
Subject: [Freeswitch-users] Anyone in UK want a
	freeUTStarcomF1000Gwifiphone?
References: <AANLkTikXhMIE8ZYXnxzO6faAXcCAt6aTYrBDbbRbqMw_@mail.gmail.com><87E1EE0F617A4C7482DCB5ABEE19E8CA@bp1.ad.bp.com><AANLkTikUqB-JP8Pp9zVisC14W24ooYkExIhHYbeFMf4a@mail.gmail.com><7A0AFD42891D42CA8D1C7F2B9A75731C@bp1.ad.bp.com><AANLkTin5V4EQEDbru3QaqJ3n69cK5nETdZxbybZAp8fW@mail.gmail.com><551DE642D9DD49D2BF55F44997822BB6@bp1.ad.bp.com>
	<AANLkTinnpxpxU_sgo4B-rTmmsHU-Y7r_KKAvf6vf4-3a@mail.gmail.com>
Message-ID: <58C7C92203DD45E3A1549F101B32624F@bp1.ad.bp.com>

te-he - nice one !
  ----- Original Message ----- 
  From: Bruce Hopkins 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Sunday, May 09, 2010 9:32 PM
  Subject: Re: [Freeswitch-users] Anyone in UK want a freeUTStarcomF1000Gwifiphone?


  I guess it's a case of he who caffeinates is lost ;-)


  On 9 May 2010 21:16, Dave Stevenson <stevendt at primrosebank.net> wrote:

    Yeah - I was just too slow !

    I knew I shouldn't have gone and made that coffee :-)

      ----- Original Message ----- 
      From: Bruce Hopkins 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Sunday, May 09, 2010 8:58 PM
      Subject: Re: [Freeswitch-users] Anyone in UK want a free UTStarcomF1000Gwifiphone?


      Sorry mate.   It was true - just fleetingly so !


      On 9 May 2010 20:48, Dave Stevenson <stevendt at primrosebank.net> wrote:

        OK, thanks anyway,

        I thought it was too good to be true :-(

        regards
        Dave
          ----- Original Message ----- 
          From: Bruce Hopkins 
          To: freeswitch-users at lists.freeswitch.org 
          Sent: Sunday, May 09, 2010 8:44 PM
          Subject: Re: [Freeswitch-users] Anyone in UK want a free UTStarcom F1000Gwifiphone?


          Dave - you were beaten to it by 1 minute I'm afraid.

          Sorry
          Bruce


          On 9 May 2010 20:31, Dave Stevenson <stevendt at primrosebank.net> wrote:

            Wow - if this is a serious offer - then I'm your man ! 

            Please please please ?

            how do we get in touch ?

            regards
            Dave


              ----- Original Message ----- 
              From: Bruce Hopkins 
              To: freeswitch-users at lists.freeswitch.org 
              Sent: Sunday, May 09, 2010 8:19 PM
              Subject: [Freeswitch-users] Anyone in UK want a free UTStarcom F1000G wifiphone?


              It's as new, in its original packaging - but I'm not going to need it and it's too obscure to sell through ebay.

              So if you want it, it's yours.

              I'm moving house and need to cut down on the amount of unused stuff here - so I'll even pay the UK postage to avoid throwing it away.

              Any taker? Seriously !

              Best regards
              Bruce



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From sos at sokhapkin.dyndns.org  Sun May  9 14:01:03 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Sun, 9 May 2010 17:01:03 -0400
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <2267B338-D811-43B0-AC98-8DDC7A8480A8@freeswitch.org>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<201005091510.46511.sos@sokhapkin.dyndns.org>
	<2267B338-D811-43B0-AC98-8DDC7A8480A8@freeswitch.org>
Message-ID: <201005091701.03473.sos@sokhapkin.dyndns.org>

Shouldn't label "end:" be BEFORE  switch_odbc_statement_handle_free(&stmt); ? 
I think it should...


static switch_status_t bill_event(
....

       if (switch_odbc_handle_exec(globals.master_odbc, sql, &stmt, NULL) != 
SWITCH_ODBC_SUCCESS) {
                char *err_str;
                err_str = switch_odbc_handle_get_error(globals.master_odbc, 
stmt);
                switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "ERR: 
[%s]\n[%s]\n", sql, switch_str_nil(err_str));
                switch_safe_free(err_str);
        } else {
                /* TODO: Failover to a flat/text file if DB is unavailable */

                goto end;
        }

        switch_odbc_statement_handle_free(&stmt);

end:


On Sunday 09 May 2010, Brian West wrote:
> What are you calling significant memory usage?
> 
> /b
> 
> On May 9, 2010, at 2:10 PM, Sergey Okhapkin wrote:
> > Valgrind output shows no significant leaks. This was discussed already in
> > this thread a month ago.
> >
> > On Sunday 09 May 2010, Brian West wrote:
> >> Please use valgrind and see where its leaking then open a jira.
> >>
> >> Nobody has been able to reproduce this in a lab nor provide any details
> >> to assist in finding the issue... All I have seen is people complaining
> >> about it and not doing what they should debugging the issue and
> >> reporting it.
> >>
> >> /b
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From bwibowo at gmail.com  Sun May  9 15:06:34 2010
From: bwibowo at gmail.com (budi wibowo)
Date: Mon, 10 May 2010 05:06:34 +0700
Subject: [Freeswitch-users] outgoing gateway
In-Reply-To: <0EC75EB8-42A9-4C56-9DA7-27C8AA642131@gmail.com>
References: <l2o547ec34a1005061642y37cd9880tac6e3aef1da1be1@mail.gmail.com>
	<1077559925-1273204385-cardhu_decombobulator_blackberry.rim.net-1446324892-@bda057.bisx.prodap.on.blackberry>
	<472BC89E-11F0-49C2-8EA5-23783BC38770@jerris.com>
	<z2y547ec34a1005071904mcc2aa5d6wbec359c34d639f61@mail.gmail.com>
	<t2t23f91031005071926l451137ffo76d667030f41f573@mail.gmail.com>
	<y2z23f91031005071930p3f96278aj9f7db3a5e0df60aa@mail.gmail.com>
	<r2h547ec34a1005080126i9e6f0cche81e1c1118ece6ef@mail.gmail.com>
	<09D47C25-81C0-4815-B477-7595D769EF60@gmail.com>
	<h2s547ec34a1005081716xc09243cq20ec07fda9ba863c@mail.gmail.com>
	<0EC75EB8-42A9-4C56-9DA7-27C8AA642131@gmail.com>
Message-ID: <v2o547ec34a1005091506m991f371aj9213eb5e4965ea80@mail.gmail.com>

thx a lot,
refering to http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start
with required changes but still failed, i also attached the F8 capture, hope
can solve this problem


regards
budi

<http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start>

On Sun, May 9, 2010 at 4:49 PM, David Ponzone <david.ponzone at gmail.com>wrote:

> You need to add this in your gateway config:
>
>   <param name="username" value="none"/>
>   <param name="password" value="none"/>
>
> and change the register param to:
>           <param name="register" value="false"/>
>
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 09/05/2010 ? 02:16, budi wibowo a ?crit :
>
> thx a lot for your explanation, i have another question, my sip server
> (mera sip-hit) dont need user name and password to connect. for security i
> just use firewall rules.
> if i refer to other samples like fwd,voicheap etc all require user name and
> password.
>
> my conf/sip_profiles/external/test.xml look like this
>  <include>
>         <gateway name="test">
>           <param name="realm" value="202.xx.xx.xx.xx"/>
>           <param name="proxy" value="202.xx.xx.xx.xx"/>
>           <param name="expire-seconds" value="3600"/>
>           <param name="register" value="true"/>
>           <param name="retry-seconds" value="3600"/>
>         </gateway>
> </include>
>
>
> my conf/dialplan/default/00_test.xml look like
>
> <include>
>    <extension name="test">
>      <condition field="destination_number" expression="^(\d)$">
>        <action application="set"
> data="effective_caller_id_number=${outbound_caller_id_number}"/>
>        <action application="set"
> data="effective_caller_id_name=${outbound_caller_id_name}"/>
>        <action application="bridge" data="sofia/gateway/test/$1"/>
>      </condition>
>    </extension>
> </include>
>
>
> but still i cant see any call coming from my mera siphit.
>
> TIA
>
> budi wibowo
>
>
>
> On Sat, May 8, 2010 at 4:04 PM, David Ponzone <david.ponzone at gmail.com>wrote:
>
>> Not sure because your logs are toon short, but it sems that your call is
>> routed to ENUM.
>> In the default config, ENUM routing is the last resort rule.
>> You need to add your own stuff before it.
>> To do so, you may add your extensions in a xml file called
>> 01_something.xml in conf/dialplan/default
>> (see 01_example.com.xml for an example).
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 08/05/2010 ? 10:26, budi wibowo a ?crit :
>>
>> dear all i try to make some changes but my call still failing
>> why the call never reach gateway i define in sip_profile/external/ and
>> call goes to voiprakyar.or.id that i never define
>>
>>
>>
>>
>> 2010-05-08 15:22:46.403733 [NOTICE] switch_channel.c:602 New Channel
>> sofia/internal/budi at sip1.xxx.com [7e483bd8-ba9d-4f1e-8560-74edb8fba2af]
>> 2010-05-08 15:22:46.407056 [INFO] mod_dialplan_xml.c:315 Processing
>> budi->62815145150 in context default
>> 2010-05-08 15:22:46.410565 [NOTICE] switch_ivr.c:1349 Transfer
>> sofia/internal/budi at sip1.xxx.com to enum[62815145150 at default]
>> 2010-05-08 15:22:46.750534 [NOTICE] switch_channel.c:602 New Channel
>> sofia/internal/62815145150 at voiprakyat.or.id[1e55a5c7-5d58-4437-a48b-4278154b57a0]
>> 2010-05-08 15:22:48.426621 [NOTICE] sofia.c:3849 Hangup sofia/internal/
>> 62815145150 at voiprakyat.or.id [CS_CONSUME_MEDIA] [CALL_REJECTED]
>>
>>
>> TIA
>>
>> budi
>>
>> On Sat, May 8, 2010 at 9:30 AM, Seven Du <dujinfang at gmail.com> wrote:
>>
>>> In the mean time, I think it's a good article for you:
>>>
>>> http://www.linuxpromagazine.com/Issues/2009/106/TALK-SOFT
>>>
>>> 2010/5/8 Seven Du <dujinfang at gmail.com>:
>>> > default configuration files of mod_sofia is in conf/sip_profiles
>>> >
>>> > http://wiki.freeswitch.org/wiki/Sofia
>>> > http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files
>>> >
>>> > 2010/5/8 budi wibowo <bwibowo at gmail.com>:
>>> >> thx, after reading the document i still got confusion about sofia
>>> module
>>> >> i give example below
>>> >>
>>> >> Dialing Through A Gateway(SIP Provider)
>>> >>
>>> >> A gateway is a means for making outbound calls through a SIP provider.
>>> For
>>> >> example:
>>> >>
>>> >> sofia/gateway/mygateway.com/1234
>>> >>
>>> >> by default i dont find any directory named sofia, where i should put
>>> this
>>> >> directory?
>>> >> many document i read telling about sofia
>>> >> TIA
>>> >> budi
>>> >>
>>> >> On Fri, May 7, 2010 at 12:55 PM, Michael Jerris <mike at jerris.com>
>>> wrote:
>>> >>>
>>> >>> http://wiki.freeswitch.org/wiki/Dialplan
>>> >>> On May 6, 2010, at 11:53 PM, Budi wibowo wrote:
>>> >>>
>>> >>> Yes I have siphit installed, I tried to make some changes on dialplan
>>> file
>>> >>> but call always goes to other server, what should I configure to
>>> implement
>>> >>> this outgoing call
>>> >>>
>>> >>>
>>> >>>
>>> >>> _______________________________________________
>>> >>> FreeSWITCH-users mailing list
>>> >>> FreeSWITCH-users at lists.freeswitch.org
>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>> http://www.freeswitch.org
>>> >>>
>>> >>
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >>
>>> >>
>>> >
>>> >
>>> >
>>> > --
>>> > Blog: http://www.dujinfang.com
>>> > Proj:  http://www.freeswitch.org.cn
>>> >
>>>
>>>
>>>
>>> --
>>> Blog: http://www.dujinfang.com
>>> Proj:  http://www.freeswitch.org.cn
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bwibowo at gmail.com  Sun May  9 15:42:24 2010
From: bwibowo at gmail.com (budi wibowo)
Date: Mon, 10 May 2010 05:42:24 +0700
Subject: [Freeswitch-users] outgoing gateway
In-Reply-To: <v2o547ec34a1005091506m991f371aj9213eb5e4965ea80@mail.gmail.com>
References: <l2o547ec34a1005061642y37cd9880tac6e3aef1da1be1@mail.gmail.com>
	<472BC89E-11F0-49C2-8EA5-23783BC38770@jerris.com>
	<z2y547ec34a1005071904mcc2aa5d6wbec359c34d639f61@mail.gmail.com>
	<t2t23f91031005071926l451137ffo76d667030f41f573@mail.gmail.com>
	<y2z23f91031005071930p3f96278aj9f7db3a5e0df60aa@mail.gmail.com>
	<r2h547ec34a1005080126i9e6f0cche81e1c1118ece6ef@mail.gmail.com>
	<09D47C25-81C0-4815-B477-7595D769EF60@gmail.com>
	<h2s547ec34a1005081716xc09243cq20ec07fda9ba863c@mail.gmail.com>
	<0EC75EB8-42A9-4C56-9DA7-27C8AA642131@gmail.com>
	<v2o547ec34a1005091506m991f371aj9213eb5e4965ea80@mail.gmail.com>
Message-ID: <o2y547ec34a1005091542p219ff4c6w5b2d64041f511e85@mail.gmail.com>

dear all
thx for your support it's working now :)
next question is after i change some xml file then i do reloadxml, but in
fact not all config changes.
reason of my statement is: many time i do reload xml but seems my config
still not work, then i shutdown the FS, then invoke ./freeswitch again and
work
any advice?


regards

budi

On Mon, May 10, 2010 at 5:06 AM, budi wibowo <bwibowo at gmail.com> wrote:

> thx a lot,
> refering to http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start
> with required changes but still failed, i also attached the F8 capture,
> hope can solve this problem
>
>
> regards
> budi
>
> <http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start>
>
> On Sun, May 9, 2010 at 4:49 PM, David Ponzone <david.ponzone at gmail.com>wrote:
>
>> You need to add this in your gateway config:
>>
>>   <param name="username" value="none"/>
>>   <param name="password" value="none"/>
>>
>> and change the register param to:
>>           <param name="register" value="false"/>
>>
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 09/05/2010 ? 02:16, budi wibowo a ?crit :
>>
>> thx a lot for your explanation, i have another question, my sip server
>> (mera sip-hit) dont need user name and password to connect. for security i
>> just use firewall rules.
>> if i refer to other samples like fwd,voicheap etc all require user name
>> and password.
>>
>> my conf/sip_profiles/external/test.xml look like this
>>  <include>
>>         <gateway name="test">
>>           <param name="realm" value="202.xx.xx.xx.xx"/>
>>           <param name="proxy" value="202.xx.xx.xx.xx"/>
>>           <param name="expire-seconds" value="3600"/>
>>           <param name="register" value="true"/>
>>           <param name="retry-seconds" value="3600"/>
>>         </gateway>
>> </include>
>>
>>
>> my conf/dialplan/default/00_test.xml look like
>>
>> <include>
>>    <extension name="test">
>>      <condition field="destination_number" expression="^(\d)$">
>>        <action application="set"
>> data="effective_caller_id_number=${outbound_caller_id_number}"/>
>>        <action application="set"
>> data="effective_caller_id_name=${outbound_caller_id_name}"/>
>>        <action application="bridge" data="sofia/gateway/test/$1"/>
>>      </condition>
>>    </extension>
>> </include>
>>
>>
>> but still i cant see any call coming from my mera siphit.
>>
>> TIA
>>
>> budi wibowo
>>
>>
>>
>> On Sat, May 8, 2010 at 4:04 PM, David Ponzone <david.ponzone at gmail.com>wrote:
>>
>>> Not sure because your logs are toon short, but it sems that your call is
>>> routed to ENUM.
>>> In the default config, ENUM routing is the last resort rule.
>>> You need to add your own stuff before it.
>>> To do so, you may add your extensions in a xml file called
>>> 01_something.xml in conf/dialplan/default
>>> (see 01_example.com.xml for an example).
>>>
>>>  David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>> l'exp?diteur.*
>>> *
>>> *
>>>
>>>
>>>
>>> Le 08/05/2010 ? 10:26, budi wibowo a ?crit :
>>>
>>> dear all i try to make some changes but my call still failing
>>> why the call never reach gateway i define in sip_profile/external/ and
>>> call goes to voiprakyar.or.id that i never define
>>>
>>>
>>>
>>>
>>> 2010-05-08 15:22:46.403733 [NOTICE] switch_channel.c:602 New Channel
>>> sofia/internal/budi at sip1.xxx.com [7e483bd8-ba9d-4f1e-8560-74edb8fba2af]
>>> 2010-05-08 15:22:46.407056 [INFO] mod_dialplan_xml.c:315 Processing
>>> budi->62815145150 in context default
>>> 2010-05-08 15:22:46.410565 [NOTICE] switch_ivr.c:1349 Transfer
>>> sofia/internal/budi at sip1.xxx.com to enum[62815145150 at default]
>>> 2010-05-08 15:22:46.750534 [NOTICE] switch_channel.c:602 New Channel
>>> sofia/internal/62815145150 at voiprakyat.or.id[1e55a5c7-5d58-4437-a48b-4278154b57a0]
>>> 2010-05-08 15:22:48.426621 [NOTICE] sofia.c:3849 Hangup sofia/internal/
>>> 62815145150 at voiprakyat.or.id [CS_CONSUME_MEDIA] [CALL_REJECTED]
>>>
>>>
>>> TIA
>>>
>>> budi
>>>
>>> On Sat, May 8, 2010 at 9:30 AM, Seven Du <dujinfang at gmail.com> wrote:
>>>
>>>> In the mean time, I think it's a good article for you:
>>>>
>>>> http://www.linuxpromagazine.com/Issues/2009/106/TALK-SOFT
>>>>
>>>> 2010/5/8 Seven Du <dujinfang at gmail.com>:
>>>> > default configuration files of mod_sofia is in conf/sip_profiles
>>>> >
>>>> > http://wiki.freeswitch.org/wiki/Sofia
>>>> > http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files
>>>> >
>>>> > 2010/5/8 budi wibowo <bwibowo at gmail.com>:
>>>> >> thx, after reading the document i still got confusion about sofia
>>>> module
>>>> >> i give example below
>>>> >>
>>>> >> Dialing Through A Gateway(SIP Provider)
>>>> >>
>>>> >> A gateway is a means for making outbound calls through a SIP
>>>> provider. For
>>>> >> example:
>>>> >>
>>>> >> sofia/gateway/mygateway.com/1234
>>>> >>
>>>> >> by default i dont find any directory named sofia, where i should put
>>>> this
>>>> >> directory?
>>>> >> many document i read telling about sofia
>>>> >> TIA
>>>> >> budi
>>>> >>
>>>> >> On Fri, May 7, 2010 at 12:55 PM, Michael Jerris <mike at jerris.com>
>>>> wrote:
>>>> >>>
>>>> >>> http://wiki.freeswitch.org/wiki/Dialplan
>>>> >>> On May 6, 2010, at 11:53 PM, Budi wibowo wrote:
>>>> >>>
>>>> >>> Yes I have siphit installed, I tried to make some changes on
>>>> dialplan file
>>>> >>> but call always goes to other server, what should I configure to
>>>> implement
>>>> >>> this outgoing call
>>>> >>>
>>>> >>>
>>>> >>>
>>>> >>> _______________________________________________
>>>> >>> FreeSWITCH-users mailing list
>>>> >>> FreeSWITCH-users at lists.freeswitch.org
>>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >>> http://www.freeswitch.org
>>>> >>>
>>>> >>
>>>> >>
>>>> >> _______________________________________________
>>>> >> FreeSWITCH-users mailing list
>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> http://www.freeswitch.org
>>>> >>
>>>> >>
>>>> >
>>>> >
>>>> >
>>>> > --
>>>> > Blog: http://www.dujinfang.com
>>>> > Proj:  http://www.freeswitch.org.cn
>>>> >
>>>>
>>>>
>>>>
>>>> --
>>>> Blog: http://www.dujinfang.com
>>>> Proj:  http://www.freeswitch.org.cn
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From jason at jasonjgw.net  Sun May  9 17:18:16 2010
From: jason at jasonjgw.net (Jason White)
Date: Mon, 10 May 2010 10:18:16 +1000
Subject: [Freeswitch-users] outgoing gateway
In-Reply-To: <o2y547ec34a1005091542p219ff4c6w5b2d64041f511e85@mail.gmail.com>
References: <472BC89E-11F0-49C2-8EA5-23783BC38770@jerris.com>
	<z2y547ec34a1005071904mcc2aa5d6wbec359c34d639f61@mail.gmail.com>
	<t2t23f91031005071926l451137ffo76d667030f41f573@mail.gmail.com>
	<y2z23f91031005071930p3f96278aj9f7db3a5e0df60aa@mail.gmail.com>
	<r2h547ec34a1005080126i9e6f0cche81e1c1118ece6ef@mail.gmail.com>
	<09D47C25-81C0-4815-B477-7595D769EF60@gmail.com>
	<h2s547ec34a1005081716xc09243cq20ec07fda9ba863c@mail.gmail.com>
	<0EC75EB8-42A9-4C56-9DA7-27C8AA642131@gmail.com>
	<v2o547ec34a1005091506m991f371aj9213eb5e4965ea80@mail.gmail.com>
	<o2y547ec34a1005091542p219ff4c6w5b2d64041f511e85@mail.gmail.com>
Message-ID: <20100510001816.GA4318@jdc.jasonjgw.net>

budi wibowo <bwibowo at gmail.com> wrote:
> dear all
> thx for your support it's working now :)
> next question is after i change some xml file then i do reloadxml, but in
> fact not all config changes.

It isn't supposed to.
> reason of my statement is: many time i do reload xml but seems my config
> still not work, then i shutdown the FS, then invoke ./freeswitch again and
> work
> any advice?

If you're changing the SIP configuration, try sofia profile <profile-name>
rescan reloadxml
or if that isn't enough, sofia profile <profile-name> restart reloadxml

Reloadxml just loads the new configuration files; it does *not* re-initialize
all of the configuration parameters.



From infos at madovsky.org  Sun May  9 18:01:21 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 9 May 2010 21:01:21 -0400
Subject: [Freeswitch-users] mod_commands
Message-ID: <72EE90044B784D0BA93BC64AC409CB1F@MOBILEE1705>

Hi,

I d like to do this :

uuid_send_dtmf ${uuid_getvar( 234569 signal_bond)} 1 at 100

but don't find the correct syntax...

Thanks for your help

F
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From brian at freeswitch.org  Sun May  9 18:05:52 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 9 May 2010 20:05:52 -0500
Subject: [Freeswitch-users] mod_commands
In-Reply-To: <72EE90044B784D0BA93BC64AC409CB1F@MOBILEE1705>
References: <72EE90044B784D0BA93BC64AC409CB1F@MOBILEE1705>
Message-ID: <424F734D-AADA-46DA-AB3D-E6426D9A0AB0@freeswitch.org>

Are you doing this at the cli?

/b

On May 9, 2010, at 8:01 PM, Madovsky wrote:

> Hi,
>  
> I d like to do this :
>  
> uuid_send_dtmf ${uuid_getvar( 234569 signal_bond)} 1 at 100
>  
> but don't find the correct syntax...
>  
> Thanks for your help
>  
> F
> _______

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From tony.tin at noahmedia.com.hk  Sun May  9 19:37:58 2010
From: tony.tin at noahmedia.com.hk (Tony Tin)
Date: Mon, 10 May 2010 10:37:58 +0800
Subject: [Freeswitch-users] short ringback tone when answer
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C5576CE4DC7@cooper>
References: <j2g191e3b11005090616ndd0b0ef8g37f60c1f1016ab4f@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5576CE4DC7@cooper>
Message-ID: <y2w191e3b11005091937k6ea8d442i798b95555e11e817@mail.gmail.com>

Hi Peter,

Thanks for your reply.

The other end is the Telco, I've confirmed with them that they don't play
the tone for us, also I got the Asterisk install in the same box, it doesn't
play the tone. The tone is played exactly when the answer function is
executed, if I put sleep 10 seconds before the answer, the tone will delay
10 seconds too, any idea?

Regards,
Tony


On Mon, May 10, 2010 at 2:33 AM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> The equipment connected in the other en of the T1 is probably playing these
> tones for you, 500ms is probably the time it takes for the called to be
> connected successfully.
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [
> freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Tony Tin [
> tony.tin at noahmedia.com.hk]
> Skickat: den 9 maj 2010 15:16
> Till: freeswitch-users at lists.freeswitch.org
> ?mne: [Freeswitch-users] short ringback tone when answer
>
> Hi All,
>
> I've just setup a new IVRS with FreeSwitch and a Digium TE220 card (which
> is connected to a 4ess T1). Every thing works fine so far except one little
> problem that annoys me. Every time I dial in the ivrs, there is a very short
> (0.5 second) ringback tone exactly while the "<action
> application="answer"/>" line is executed in the dial plan. I want to disable
> this ringback tone because the ivrs will answer the call immediately, the
> ringback tone is really unnecessary in this case, but I fail to do that. I
> try to set the variable "ringback" to change the ringback tone, it's also
> not working. Could anyone please help.
>
> I'm using the native pri stack.
> The default.xml containts only below lines, and attached is the log file.
>
>       <extension name="testing">
>          <condition field="destination_number" expression="^(6.*)$">
>                <action application="sleep" data="3000"/>
>                <action application="answer"/>
>                <action application="sleep" data="3000"/>
>          </condition>
>       </extension>
>
> Regards,
> Tony
>
> !DSPAM:4be6f37432931620317181!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From infos at madovsky.org  Sun May  9 19:53:54 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 9 May 2010 22:53:54 -0400
Subject: [Freeswitch-users] mod_commands
References: <72EE90044B784D0BA93BC64AC409CB1F@MOBILEE1705>
	<424F734D-AADA-46DA-AB3D-E6426D9A0AB0@freeswitch.org>
Message-ID: <0734AD76FB8C4E159C9E483D89EAD7C6@MOBILEE1705>

yes
  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Sunday, May 09, 2010 9:05 PM
  Subject: Re: [Freeswitch-users] mod_commands


  Are you doing this at the cli?


  /b


  On May 9, 2010, at 8:01 PM, Madovsky wrote:


    Hi,

    I d like to do this :

    uuid_send_dtmf ${uuid_getvar( 234569 signal_bond)} 1 at 100

    but don't find the correct syntax...

    Thanks for your help

    F
    _______




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


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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From brian at freeswitch.org  Sun May  9 20:13:23 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 9 May 2010 22:13:23 -0500
Subject: [Freeswitch-users] mod_commands
In-Reply-To: <0734AD76FB8C4E159C9E483D89EAD7C6@MOBILEE1705>
References: <72EE90044B784D0BA93BC64AC409CB1F@MOBILEE1705>
	<424F734D-AADA-46DA-AB3D-E6426D9A0AB0@freeswitch.org>
	<0734AD76FB8C4E159C9E483D89EAD7C6@MOBILEE1705>
Message-ID: <E6AD8B0B-019A-4741-ABAE-0815BDF9628E@freeswitch.org>

prefix it with 'expand'

Sent from my iPad

On May 9, 2010, at 9:53 PM, "Madovsky" <infos at madovsky.org> wrote:

> yes
> ----- Original Message -----
> From: Brian West
> To: freeswitch-users at lists.freeswitch.org
> Sent: Sunday, May 09, 2010 9:05 PM
> Subject: Re: [Freeswitch-users] mod_commands
> 
> Are you doing this at the cli?
> 
> /b
> 
> On May 9, 2010, at 8:01 PM, Madovsky wrote:
> 
>> Hi,
>>  
>> I d like to do this :
>>  
>> uuid_send_dtmf ${uuid_getvar( 234569 signal_bond)} 1 at 100
>>  
>> but don't find the correct syntax...
>>  
>> Thanks for your help
>>  
>> F
>> _______
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From infos at madovsky.org  Sun May  9 20:35:42 2010
From: infos at madovsky.org (Madovsky)
Date: Sun, 9 May 2010 23:35:42 -0400
Subject: [Freeswitch-users] mod_commands
Message-ID: <2135DB17ADA94009B13D7BDC8E30E181@MOBILEE1705>

found :

expand uuid_send_dtmf ${uuid_getvar(4345345 signal_bond)} 2 at 400

no  pain anymore :)


  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Sunday, May 09, 2010 10:53 PM
  Subject: Re: [Freeswitch-users] mod_commands


  yes
    ----- Original Message ----- 
    From: Brian West 
    To: freeswitch-users at lists.freeswitch.org 
    Sent: Sunday, May 09, 2010 9:05 PM
    Subject: Re: [Freeswitch-users] mod_commands


    Are you doing this at the cli? 


    /b


    On May 9, 2010, at 8:01 PM, Madovsky wrote:


      Hi,

      I d like to do this :

      uuid_send_dtmf ${uuid_getvar( 234569 signal_bond)} 1 at 100

      but don't find the correct syntax...

      Thanks for your help

      F
      _______




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


    _______________________________________________
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    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From babak.freeswitch at gmail.com  Sun May  9 22:28:38 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Mon, 10 May 2010 09:58:38 +0430
Subject: [Freeswitch-users] listening on multiple events in mod_managed
Message-ID: <AANLkTik0oTbEHStsteb0n4RX8bian_CG_r-5qJxKLYN1@mail.gmail.com>

Hi
How is it possible to listen for maore than one type of events but not all
of them something like
EventConsumer con = new EventConsumer("CUSTOM CHANNEL_DESTROY
CHANNEL_HANGUP", null);
thanks
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From grsingh750 at gmail.com  Sun May  9 22:36:54 2010
From: grsingh750 at gmail.com (guru singh)
Date: Mon, 10 May 2010 11:06:54 +0530
Subject: [Freeswitch-users] group_confirm_file
Message-ID: <h2t7b463d951005092236l38e0695evbafeb218df7e1545@mail.gmail.com>

Hi,
I am using group_confirm_file to play a message to the called party...
I am not being able to sync it properly, i want it to start playing as
soon as the phone is answred... right now the called party hears it
from a random point depending upon how soon the call was answered i
guess.
How do I manage this?

here is my config file http://pastebin.freeswitch.org/12943

thanks
gs


From brian at freeswitch.org  Sun May  9 22:50:55 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 10 May 2010 00:50:55 -0500
Subject: [Freeswitch-users] group_confirm_file
In-Reply-To: <h2t7b463d951005092236l38e0695evbafeb218df7e1545@mail.gmail.com>
References: <h2t7b463d951005092236l38e0695evbafeb218df7e1545@mail.gmail.com>
Message-ID: <85D80CC0-1456-4C96-835E-9DE736D244EE@freeswitch.org>

ignore_early_media=true variable.

/b

Sent from my iPad

On May 10, 2010, at 12:36 AM, guru singh <grsingh750 at gmail.com> wrote:

> Hi,
> I am using group_confirm_file to play a message to the called party...
> I am not being able to sync it properly, i want it to start playing as
> soon as the phone is answred... right now the called party hears it
> from a random point depending upon how soon the call was answered i
> guess.
> How do I manage this?
> 
> here is my config file http://pastebin.freeswitch.org/12943
> 
> thanks
> gs
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From helmut.kuper at ewetel.de  Mon May 10 02:34:42 2010
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Mon, 10 May 2010 11:34:42 +0200
Subject: [Freeswitch-users] bridge_hangup_cause not for internal sip calls?
Message-ID: <4BE7D332.6060302@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello,

in my dialplan for calls from internal phones I transfer each call after
hangup to a dialplan which handles the hangup causes (e.g. for
announcements). Unfortunately there are at least two different channel
variables used to signal the hangup cause back to FS depending on what
type of target/interface was dialed.

When I dial an busy internal sip phone, the hangup cause can be found in
"originate_disposition"

When I dial an busy external freetdm target, the hangup cause can be
found in "bridge_hangup_cause" and "originate_disposition" is set to
"SUCCESS".

Is the a consolidated single channel variable for all types of interfaces?


regards
helmut
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From helmut.kuper at ewetel.de  Mon May 10 04:42:10 2010
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Mon, 10 May 2010 13:42:10 +0200
Subject: [Freeswitch-users] call recording with Snom and SIP-INFO not working
Message-ID: <4BE7F112.3030008@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello,

I found that call recording using Snoms which sending a SIP-INFO with
"Record: on" leads to a disconnect.

My sip-Profile has arecord-template defined:

<param name="record-template"
value="recordings/${caller_id_number}.${destination_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>

AFAIK this is the default one.

My dialplan hast the following bind-meta-app:

<action application="export" data="RECORD_STEREO=true"/>
<action application="bind_meta_app" data="9 ab s
record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>

$base_dir is "/opt/app/voip/ippbx"

"recordings"-directory exists.

When I press the record button on my Snom, it sends a SIP-INFO
containing a "Record: on" entry. FS detects this and disconnects the call.

FS's logfile shows this:

[...]
EXECUTE sofia/internal/2850 at 85.16.246.6 bind_meta_app(9 ab s
record_session::/opt/app/voip/ippbx/recordings/2850.2010-05-10-13-24-37.wav)
2010-05-10 13:24:37.257965 [INFO] switch_ivr_async.c:2332 Bound A-Leg: 9
record_session::/opt/app/voip/ippbx/recordings/2850.2010-05-10-13-24-37.wav
2010-05-10 13:24:37.257965 [INFO] switch_ivr_async.c:2339 Bound B-Leg: 9
record_session::/opt/app/voip/ippbx/recordings/2850.2010-05-10-13-24-37.wav
[...]

Sending SIP-INFO from Snom now

2010-05-10 13:24:40.157036 [ERR] mod_sndfile.c:195 Error Opening File
[/opt/app/voip/ippbx/recordings//opt/app/voip/ippbx/recordings/2850.4916.2010-05-10-13-24-40.wav]
[System error : Datei oder Verzeichnis nicht gefunden.]
2010-05-10 13:24:40.157036 [ERR] switch_ivr_async.c:997 Error opening
/opt/app/voip/ippbx/recordings//opt/app/voip/ippbx/recordings/2850.4916.2010-05-10-13-24-40.wav
2010-05-10 13:24:40.157036 [NOTICE] switch_ivr_async.c:998 Hangup
sofia/internal/2850 at 85.16.246.6 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER]
[...]

So it's quite clear what happens here: Somehow FS creates a wrong file
path ("/opt/app/voip/ippbx/recordings//opt/app/voip/ippbx/" instead of
"/opt/app/voip/ippbx/recordings/"]


When I do the same by using "*9" on my Snom, everything works fine.

I'm using "FreeSWITCH Version 1.0.head (17097:17188M)"

Any ideas what I'm doing wrong?
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From bwibowo at gmail.com  Mon May 10 05:19:07 2010
From: bwibowo at gmail.com (Budi wibowo)
Date: Mon, 10 May 2010 12:19:07 +0000
Subject: [Freeswitch-users] outgoing gateway
In-Reply-To: <20100510001816.GA4318@jdc.jasonjgw.net>
References: <472BC89E-11F0-49C2-8EA5-23783BC38770@jerris.com><z2y547ec34a1005071904mcc2aa5d6wbec359c34d639f61@mail.gmail.com><t2t23f91031005071926l451137ffo76d667030f41f573@mail.gmail.com><y2z23f91031005071930p3f96278aj9f7db3a5e0df60aa@mail.gmail.com><r2h547ec34a1005080126i9e6f0cche81e1c1118ece6ef@mail.gmail.com><09D47C25-81C0-4815-B477-7595D769EF60@gmail.com><h2s547ec34a1005081716xc09243cq20ec07fda9ba863c@mail.gmail.com><0EC75EB8-42A9-4C56-9DA7-27C8AA642131@gmail.com><v2o547ec34a1005091506m991f371aj9213eb5e4965ea80@mail.gmail.com><o2y547ec34a1005091542p219ff4c6w5b2d64041f511e85@mail.gmail.com><20100510001816.GA4318@jdc.jasonjgw.net>
Message-ID: <1419115537-1273493954-cardhu_decombobulator_blackberry.rim.net-840701352-@bda057.bisx.prodap.on.blackberry>

Thx again, now I move to cdr part. Both cdr in csv and xml are working, but I want to store in mysql db. Db structure already created, following the wiki doc for mod_cdr. But I don't see the cdr logged into mysql, pressing F8 also not showing any db query. 
Any help?
Mod_cdr loaded via modules.conf.xml 
TIA
Budi wibowo

-----Original Message-----
From: Jason White <jason at jasonjgw.net>
Date: Mon, 10 May 2010 10:18:16 
To: <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] outgoing gateway

budi wibowo <bwibowo at gmail.com> wrote:
> dear all
> thx for your support it's working now :)
> next question is after i change some xml file then i do reloadxml, but in
> fact not all config changes.

It isn't supposed to.
> reason of my statement is: many time i do reload xml but seems my config
> still not work, then i shutdown the FS, then invoke ./freeswitch again and
> work
> any advice?

If you're changing the SIP configuration, try sofia profile <profile-name>
rescan reloadxml
or if that isn't enough, sofia profile <profile-name> restart reloadxml

Reloadxml just loads the new configuration files; it does *not* re-initialize
all of the configuration parameters.


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

From brian at freeswitch.org  Mon May 10 06:58:48 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 10 May 2010 08:58:48 -0500
Subject: [Freeswitch-users] call recording with Snom and SIP-INFO not
	working
In-Reply-To: <4BE7F112.3030008@ewetel.de>
References: <4BE7F112.3030008@ewetel.de>
Message-ID: <AF191A17-D250-47A3-9B98-E329E173F037@freeswitch.org>

No its clear you didn't set the record-template and record-path correctly on the sofia profile.

<param name="record-path" value="$${recordings_dir}"/>
<param name="record-template" value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>

Please verify you set the record-path too.

/b


On May 10, 2010, at 6:42 AM, Helmut Kuper wrote:

> So it's quite clear what happens here: Somehow FS creates a wrong file
> path ("/opt/app/voip/ippbx/recordings//opt/app/voip/ippbx/" instead of
> "/opt/app/voip/ippbx/recordings/"]



From helmut.kuper at ewetel.de  Mon May 10 07:35:43 2010
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Mon, 10 May 2010 16:35:43 +0200
Subject: [Freeswitch-users] call recording with Snom and SIP-INFO not
 working
In-Reply-To: <AF191A17-D250-47A3-9B98-E329E173F037@freeswitch.org>
References: <4BE7F112.3030008@ewetel.de>
	<AF191A17-D250-47A3-9B98-E329E173F037@freeswitch.org>
Message-ID: <4BE819BF.4060909@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Brian,


yep, right. Found that too. I added "<param name="record-path"
value="$${recordings_dir}"/>" to sip-profile and removed any path
information from record-template.

I rescaned sofia profile the result was this:

2010-05-10 16:28:22.715231 [ERR] mod_sndfile.c:195 Error Opening File
[/opt/app/voip/ippbx/recordings/recordings/2850.4916.2010-05-10-16-28-22.wav]
[System error : Datei oder Verzeichnis nicht gefunden.]
2010-05-10 16:28:22.715231 [ERR] switch_ivr_async.c:997 Error opening
/opt/app/voip/ippbx/recordings/recordings/2850.4916.2010-05-10-16-28-22.wav


Then I set recordings_dir in vars.xml to "$${base_dir}/recordings" which
led to success.

Thx for your help.



On 10.05.2010 15:58, Brian West wrote:
> No its clear you didn't set the record-template and record-path correctly on the sofia profile.
> 
> <param name="record-path" value="$${recordings_dir}"/>
> <param name="record-template" value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
> 
> Please verify you set the record-path too.
> 
> /b


- -- 

Mit freundlichen Gr??en
Helmut Kuper
Gesch?ftseinheit FD - L?sungen f?r Finanzdienstleister
Telefax: (0441) 8000-2799
mailto:helmut.kuper at ewetel.de
___________________________________
EWE TEL GmbH
Cloppenburger Stra?e 310
26133 Oldenburg
EWE TEL GmbH

Handelsregister Amtsgericht Oldenburg HRB 3723
Vorsitzender des Aufsichtsrates: Dr. Werner Brinker
Gesch?ftsf?hrung: Hans-Joachim Iken (Vorsitzender),
Ulf Heggenberger, Dr. Norbert Schulz, Dirk Thole
Homepage: http://www.ewetel.de
___________________________________
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From helmut.kuper at ewetel.de  Mon May 10 08:15:49 2010
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Mon, 10 May 2010 17:15:49 +0200
Subject: [Freeswitch-users] call recording with Snom and SIP-INFO not
 working
In-Reply-To: <4BE819BF.4060909@ewetel.de>
References: <4BE7F112.3030008@ewetel.de>	<AF191A17-D250-47A3-9B98-E329E173F037@freeswitch.org>
	<4BE819BF.4060909@ewetel.de>
Message-ID: <4BE82325.9050703@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Brian,

I enhanced your wiki about this.

A further question to this: The options (a,b,o,s,i ... ) which can be
used in bind_meta_app are not accessible by this way of session recording?


regards
Helmut


On 10.05.2010 16:35, Helmut Kuper wrote:
> Hi Brian,
> 
> 
> yep, right. Found that too. I added "<param name="record-path"
> value="$${recordings_dir}"/>" to sip-profile and removed any path
> information from record-template.
> 
> I rescaned sofia profile the result was this:
> 
> 2010-05-10 16:28:22.715231 [ERR] mod_sndfile.c:195 Error Opening File
> [/opt/app/voip/ippbx/recordings/recordings/2850.4916.2010-05-10-16-28-22.wav]
> [System error : Datei oder Verzeichnis nicht gefunden.]
> 2010-05-10 16:28:22.715231 [ERR] switch_ivr_async.c:997 Error opening
> /opt/app/voip/ippbx/recordings/recordings/2850.4916.2010-05-10-16-28-22.wav
> 
> 
> Then I set recordings_dir in vars.xml to "$${base_dir}/recordings" which
> led to success.
> 
> Thx for your help.
> 
> 
> 
> On 10.05.2010 15:58, Brian West wrote:
>> No its clear you didn't set the record-template and record-path correctly on the sofia profile.
> 
>> <param name="record-path" value="$${recordings_dir}"/>
>> <param name="record-template" value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
> 
>> Please verify you set the record-path too.
> 
>> /b
> 
> 

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From anthony.minessale at gmail.com  Mon May 10 09:01:50 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 10 May 2010 11:01:50 -0500
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <201005091701.03473.sos@sokhapkin.dyndns.org>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<201005091510.46511.sos@sokhapkin.dyndns.org>
	<2267B338-D811-43B0-AC98-8DDC7A8480A8@freeswitch.org>
	<201005091701.03473.sos@sokhapkin.dyndns.org>
Message-ID: <AANLkTimTdNs40gq6WTuvCtWdrVxgcyUYgsuYt4fcZZWe@mail.gmail.com>

Why does this thread continue on the list instead of being opened in JIRA
Can you see how this issue can be forgotten because it's on the mailing list
that gets 200 emails a day?
When you show the code in the body of an email like that, then I have to
spend a minuted wondering where in the code you were talking about because
it's not in the form of a unified diff.

I appreciate that you do not normally have to deal with organized software
management but I must ask that you please spend a little time to learn our
customs and use JIRA and patches to communicate bug reports.  I am glad you
are helping to find problems but we really need to stay organized to make it
forward.

I redid the function in latest GIT.  It did look like it was doing a few
things wrong but they probably would have surfaced in valgrind if you were
running it in full leak check mode.  When you say it goes to 300M, does it
keep going from there because there is a bit of memory that will be pooled
as you start to use more advanced features.  Did you ever watch it to see
how high it will go?



On Sun, May 9, 2010 at 4:01 PM, Sergey Okhapkin <sos at sokhapkin.dyndns.org>wrote:

> Shouldn't label "end:" be BEFORE  switch_odbc_statement_handle_free(&stmt);
> ?
> I think it should...
>
>
> static switch_status_t bill_event(
> ....
>
>       if (switch_odbc_handle_exec(globals.master_odbc, sql, &stmt, NULL) !=
> SWITCH_ODBC_SUCCESS) {
>                char *err_str;
>                err_str = switch_odbc_handle_get_error(globals.master_odbc,
> stmt);
>                switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR,
> "ERR:
> [%s]\n[%s]\n", sql, switch_str_nil(err_str));
>                switch_safe_free(err_str);
>        } else {
>                /* TODO: Failover to a flat/text file if DB is unavailable
> */
>
>                goto end;
>        }
>
>        switch_odbc_statement_handle_free(&stmt);
>
> end:
>
>
> On Sunday 09 May 2010, Brian West wrote:
> > What are you calling significant memory usage?
> >
> > /b
> >
> > On May 9, 2010, at 2:10 PM, Sergey Okhapkin wrote:
> > > Valgrind output shows no significant leaks. This was discussed already
> in
> > > this thread a month ago.
> > >
> > > On Sunday 09 May 2010, Brian West wrote:
> > >> Please use valgrind and see where its leaking then open a jira.
> > >>
> > >> Nobody has been able to reproduce this in a lab nor provide any
> details
> > >> to assist in finding the issue... All I have seen is people
> complaining
> > >> about it and not doing what they should debugging the issue and
> > >> reporting it.
> > >>
> > >> /b
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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From anthony.minessale at gmail.com  Mon May 10 09:08:18 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 10 May 2010 11:08:18 -0500
Subject: [Freeswitch-users] bridge_hangup_cause not for internal sip
	calls?
In-Reply-To: <4BE7D332.6060302@ewetel.de>
References: <4BE7D332.6060302@ewetel.de>
Message-ID: <AANLkTinIevjzNaO3rj9s5vsPKNxTb6iD51rhEeHHcMcy@mail.gmail.com>

maybe your TDM is doing early media which would produce a SUCESSFUL
originate disposition since the
origination ends as soon as the call reaches early media.


On Mon, May 10, 2010 at 4:34 AM, Helmut Kuper <helmut.kuper at ewetel.de>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello,
>
> in my dialplan for calls from internal phones I transfer each call after
> hangup to a dialplan which handles the hangup causes (e.g. for
> announcements). Unfortunately there are at least two different channel
> variables used to signal the hangup cause back to FS depending on what
> type of target/interface was dialed.
>
> When I dial an busy internal sip phone, the hangup cause can be found in
> "originate_disposition"
>
> When I dial an busy external freetdm target, the hangup cause can be
> found in "bridge_hangup_cause" and "originate_disposition" is set to
> "SUCCESS".
>
> Is the a consolidated single channel variable for all types of interfaces?
>
>
> regards
> helmut
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.7 (MingW32)
>
> iD8DBQFL59Mx4tZeNddg3dwRAkwDAJ0VjInzSHT9lpc0rhSVLXTBhsPLpQCgi1VL
> EzizysQI373J/c8GCHAetzg=
> =ImNz
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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From sos at sokhapkin.dyndns.org  Mon May 10 09:14:39 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 10 May 2010 12:14:39 -0400
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <AANLkTimTdNs40gq6WTuvCtWdrVxgcyUYgsuYt4fcZZWe@mail.gmail.com>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<201005091701.03473.sos@sokhapkin.dyndns.org>
	<AANLkTimTdNs40gq6WTuvCtWdrVxgcyUYgsuYt4fcZZWe@mail.gmail.com>
Message-ID: <201005101214.39428.sos@sokhapkin.dyndns.org>

It's already on jira - FSMOD-48. Valgrind with --leak-check=full finds 
nothing.

Memory kept growing and growing, I had to restart FS when RSS reached 500M to 
avoid swap.

On Monday 10 May 2010, Anthony Minessale wrote:
> Why does this thread continue on the list instead of being opened in JIRA
> Can you see how this issue can be forgotten because it's on the mailing
>  list that gets 200 emails a day?
> When you show the code in the body of an email like that, then I have to
> spend a minuted wondering where in the code you were talking about because
> it's not in the form of a unified diff.
> 
> I appreciate that you do not normally have to deal with organized software
> management but I must ask that you please spend a little time to learn our
> customs and use JIRA and patches to communicate bug reports.  I am glad you
> are helping to find problems but we really need to stay organized to make
>  it forward.
> 
> I redid the function in latest GIT.  It did look like it was doing a few
> things wrong but they probably would have surfaced in valgrind if you were
> running it in full leak check mode.  When you say it goes to 300M, does it
> keep going from there because there is a bit of memory that will be pooled
> as you start to use more advanced features.  Did you ever watch it to see
> how high it will go?
> 
> On Sun, May 9, 2010 at 4:01 PM, Sergey Okhapkin 
<sos at sokhapkin.dyndns.org>wrote:
> > Shouldn't label "end:" be BEFORE 
> > switch_odbc_statement_handle_free(&stmt); ?
> > I think it should...
> >
> >
> > static switch_status_t bill_event(
> > ....
> >
> >       if (switch_odbc_handle_exec(globals.master_odbc, sql, &stmt, NULL)
> > != SWITCH_ODBC_SUCCESS) {
> >                char *err_str;
> >                err_str =
> > switch_odbc_handle_get_error(globals.master_odbc, stmt);
> >                switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR,
> > "ERR:
> > [%s]\n[%s]\n", sql, switch_str_nil(err_str));
> >                switch_safe_free(err_str);
> >        } else {
> >                /* TODO: Failover to a flat/text file if DB is unavailable
> > */
> >
> >                goto end;
> >        }
> >
> >        switch_odbc_statement_handle_free(&stmt);
> >
> > end:
> >
> > On Sunday 09 May 2010, Brian West wrote:
> > > What are you calling significant memory usage?
> > >
> > > /b
> > >
> > > On May 9, 2010, at 2:10 PM, Sergey Okhapkin wrote:
> > > > Valgrind output shows no significant leaks. This was discussed
> > > > already
> >
> > in
> >
> > > > this thread a month ago.
> > > >
> > > > On Sunday 09 May 2010, Brian West wrote:
> > > >> Please use valgrind and see where its leaking then open a jira.
> > > >>
> > > >> Nobody has been able to reproduce this in a lab nor provide any
> >
> > details
> >
> > > >> to assist in finding the issue... All I have seen is people
> >
> > complaining
> >
> > > >> about it and not doing what they should debugging the issue and
> > > >> reporting it.
> > > >>
> > > >> /b
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user
> > >s http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 



From anthony.minessale at gmail.com  Mon May 10 09:14:50 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 10 May 2010 11:14:50 -0500
Subject: [Freeswitch-users] short ringback tone when answer
In-Reply-To: <y2w191e3b11005091937k6ea8d442i798b95555e11e817@mail.gmail.com>
References: <j2g191e3b11005090616ndd0b0ef8g37f60c1f1016ab4f@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C5576CE4DC7@cooper>
	<y2w191e3b11005091937k6ea8d442i798b95555e11e817@mail.gmail.com>
Message-ID: <AANLkTikFk61Phm9lKYIh33BfsMFPfgFZrO-rivpJzi3v@mail.gmail.com>

if ringback is not set, it would never play a tone and even if it was, it
would not be played if you explicitly called answer.
Try reversing the answer and the sleep in your dialplan.




On Sun, May 9, 2010 at 9:37 PM, Tony Tin <tony.tin at noahmedia.com.hk> wrote:

> Hi Peter,
>
> Thanks for your reply.
>
> The other end is the Telco, I've confirmed with them that they don't play
> the tone for us, also I got the Asterisk install in the same box, it doesn't
> play the tone. The tone is played exactly when the answer function is
> executed, if I put sleep 10 seconds before the answer, the tone will delay
> 10 seconds too, any idea?
>
> Regards,
> Tony
>
>
>
> On Mon, May 10, 2010 at 2:33 AM, Peter Olsson <
> peter.olsson at visionutveckling.se> wrote:
>
>> The equipment connected in the other en of the T1 is probably playing
>> these tones for you, 500ms is probably the time it takes for the called to
>> be connected successfully.
>>
>> /Peter
>> ________________________________________
>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [
>> freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Tony Tin [
>> tony.tin at noahmedia.com.hk]
>> Skickat: den 9 maj 2010 15:16
>> Till: freeswitch-users at lists.freeswitch.org
>> ?mne: [Freeswitch-users] short ringback tone when answer
>>
>> Hi All,
>>
>> I've just setup a new IVRS with FreeSwitch and a Digium TE220 card (which
>> is connected to a 4ess T1). Every thing works fine so far except one little
>> problem that annoys me. Every time I dial in the ivrs, there is a very short
>> (0.5 second) ringback tone exactly while the "<action
>> application="answer"/>" line is executed in the dial plan. I want to disable
>> this ringback tone because the ivrs will answer the call immediately, the
>> ringback tone is really unnecessary in this case, but I fail to do that. I
>> try to set the variable "ringback" to change the ringback tone, it's also
>> not working. Could anyone please help.
>>
>> I'm using the native pri stack.
>> The default.xml containts only below lines, and attached is the log file.
>>
>>       <extension name="testing">
>>          <condition field="destination_number" expression="^(6.*)$">
>>                <action application="sleep" data="3000"/>
>>                <action application="answer"/>
>>                <action application="sleep" data="3000"/>
>>          </condition>
>>       </extension>
>>
>> Regards,
>> Tony
>>
>> !DSPAM:4be6f37432931620317181!
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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From anthony.minessale at gmail.com  Mon May 10 09:20:16 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 10 May 2010 11:20:16 -0500
Subject: [Freeswitch-users] Alarms-codes
In-Reply-To: <3E6354B851CB4C08BE3592A43F9503A5@dell9400>
References: <017a01caec8a$ee74bb70$cb5e3250$@com>
	<w2x87f2f3b91005051355j7c5c775dld43c79c6bf9d61cb@mail.gmail.com>
	<A60A2CFD3B5C41D68FB970C1EB64DFF2@bp1.ad.bp.com>
	<g2n6b65470d1005070808y30d1dce7q1c9b4fec8cc8f058@mail.gmail.com>
	<i2g87f2f3b91005071459i16d8bd2fr7fa57a6304d2c0dc@mail.gmail.com>
	<h2k23f91031005071842t865d2765x595a7a2572f13638@mail.gmail.com>
	<z2o87f2f3b91005072246y39206088g6900a8a81434626c@mail.gmail.com>
	<SNT135-w5FF0C5B6D49D392DF23CDD2F70@phx.gbl>
	<y2zc4d05cbe1005081101ma76fb71do61a7469d1990096c@mail.gmail.com>
	<3E6354B851CB4C08BE3592A43F9503A5@dell9400>
Message-ID: <AANLkTilxa0csVAzwlCeTcI6Hv9Z2sgKqmgR_jYQqtiFP@mail.gmail.com>

Moises is right.
The best place for this is in an Event or we would be duplicating things.
Also consuming events is easier than trying to scan a log file.


On Sat, May 8, 2010 at 5:01 PM, Jan Berger <jan.berger at video24.no> wrote:

>  You could add an ?Alarm? statement in coding, but error?s and warning?s
> serve the same purpose and you would in reality be coding an ?alarm? side by
> side with these.
>
>
>
> Actually ? all I really want is the module name + code. Text, severity and
> what to do we look up in a config database.
>
>
>
> jan
>
>
>  ------------------------------
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Moises Silva
> *Sent:* 8. mai 2010 20:01
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Alarms-codes
>
>
>
> Hello Jan,
>
>
>
> I believe you're confusing alarms with logging and those two are not the
> same thing and I think should not be mixed. For alarms an special event
> should be triggered ( as mentioned in other thread, may be SWITCH_EVENT_TRAP
> ).
>
>
>
> mod_freetdm ( FreeSWITCH PSTN module ) is already sending a
> SWITCH_EVENT_TRAP when any pstn port becomes alarmed, other modules may
> start using it for their own type of alarms. Then we may add the alarm-code
> and severity fields.
>
>
> Moises Silva
> Senior Software Engineer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
> 9T3 Canada
> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>
>  On Sat, May 8, 2010 at 2:47 AM, Jan Berger <janvb at live.com> wrote:
>
> hi,
>
> I have playing around with FreeSWITCHf or a while and one of things I am
> looking for is unique Alarms-codes
>
> FreeSwitch has error, warning, info etc on the logging - but the message
> print module name and line-number - it does not incude a proper, unique
> alarm-code. Using the combination of module-name and line-number is possible
> - but this changes from release to release.
>
> The point with alarms is that they are forwarded into
> databases/applications/operating senters that monitor telecom equipmenet.
> The way this usually is done is that the forwarding mechanism filter on
> alarm-code/severity as we don't want to forward junk - but it's a
> requirement that we visualise those that affect business.
>
> ---
>
> It's doable to change logging so that every module insert a unique number -
> starting with 1 - this would make modulename + alarm-code an unique
> combination that would serve the purpose.
>
> Any suggestions?
>
> Jan
>
>
>  ------------------------------
>
> Hotmail: Trusted email with Microsoft?s powerful SPAM protection. Sign up
> now. <https://signup.live.com/signup.aspx?id=60969>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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From anthony.minessale at gmail.com  Mon May 10 09:23:06 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 10 May 2010 11:23:06 -0500
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <201005101214.39428.sos@sokhapkin.dyndns.org>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<201005091701.03473.sos@sokhapkin.dyndns.org>
	<AANLkTimTdNs40gq6WTuvCtWdrVxgcyUYgsuYt4fcZZWe@mail.gmail.com>
	<201005101214.39428.sos@sokhapkin.dyndns.org>
Message-ID: <AANLkTikWXr8bdP2VV677-nhlZF7i2tLcv_wBzPRY--IF@mail.gmail.com>

ok so you don't know for sure then, this is obviously a small box.
My browser right now is using 500m to let me write this email to you.

if you find nothing, it would suggest pools growing more than a real leak
since valgrind would normally pick it up.
I did not make this nibble bill module but I did correct the code in the
place you pointed out.  Give it a try.

I wish I could nibble-bill this mailing list =p


On Mon, May 10, 2010 at 11:14 AM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> It's already on jira - FSMOD-48. Valgrind with --leak-check=full finds
> nothing.
>
> Memory kept growing and growing, I had to restart FS when RSS reached 500M
> to
> avoid swap.
>
> On Monday 10 May 2010, Anthony Minessale wrote:
> > Why does this thread continue on the list instead of being opened in JIRA
> > Can you see how this issue can be forgotten because it's on the mailing
> >  list that gets 200 emails a day?
> > When you show the code in the body of an email like that, then I have to
> > spend a minuted wondering where in the code you were talking about
> because
> > it's not in the form of a unified diff.
> >
> > I appreciate that you do not normally have to deal with organized
> software
> > management but I must ask that you please spend a little time to learn
> our
> > customs and use JIRA and patches to communicate bug reports.  I am glad
> you
> > are helping to find problems but we really need to stay organized to make
> >  it forward.
> >
> > I redid the function in latest GIT.  It did look like it was doing a few
> > things wrong but they probably would have surfaced in valgrind if you
> were
> > running it in full leak check mode.  When you say it goes to 300M, does
> it
> > keep going from there because there is a bit of memory that will be
> pooled
> > as you start to use more advanced features.  Did you ever watch it to see
> > how high it will go?
> >
> > On Sun, May 9, 2010 at 4:01 PM, Sergey Okhapkin
> <sos at sokhapkin.dyndns.org>wrote:
> > > Shouldn't label "end:" be BEFORE
> > > switch_odbc_statement_handle_free(&stmt); ?
> > > I think it should...
> > >
> > >
> > > static switch_status_t bill_event(
> > > ....
> > >
> > >       if (switch_odbc_handle_exec(globals.master_odbc, sql, &stmt,
> NULL)
> > > != SWITCH_ODBC_SUCCESS) {
> > >                char *err_str;
> > >                err_str =
> > > switch_odbc_handle_get_error(globals.master_odbc, stmt);
> > >                switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR,
> > > "ERR:
> > > [%s]\n[%s]\n", sql, switch_str_nil(err_str));
> > >                switch_safe_free(err_str);
> > >        } else {
> > >                /* TODO: Failover to a flat/text file if DB is
> unavailable
> > > */
> > >
> > >                goto end;
> > >        }
> > >
> > >        switch_odbc_statement_handle_free(&stmt);
> > >
> > > end:
> > >
> > > On Sunday 09 May 2010, Brian West wrote:
> > > > What are you calling significant memory usage?
> > > >
> > > > /b
> > > >
> > > > On May 9, 2010, at 2:10 PM, Sergey Okhapkin wrote:
> > > > > Valgrind output shows no significant leaks. This was discussed
> > > > > already
> > >
> > > in
> > >
> > > > > this thread a month ago.
> > > > >
> > > > > On Sunday 09 May 2010, Brian West wrote:
> > > > >> Please use valgrind and see where its leaking then open a jira.
> > > > >>
> > > > >> Nobody has been able to reproduce this in a lab nor provide any
> > >
> > > details
> > >
> > > > >> to assist in finding the issue... All I have seen is people
> > >
> > > complaining
> > >
> > > > >> about it and not doing what they should debugging the issue and
> > > > >> reporting it.
> > > > >>
> > > > >> /b
> > > >
> > > > _______________________________________________
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-user
> > > >s http://www.freeswitch.org
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
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>
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From jsabater at gmail.com  Mon May 10 06:25:03 2010
From: jsabater at gmail.com (Jaume Sabater)
Date: Mon, 10 May 2010 15:25:03 +0200
Subject: [Freeswitch-users] Encoding issue with mod_xml_cdr (apparently)
In-Reply-To: <1871ECF8-82DD-4DB6-9C52-C274648CDE4B@freeswitch.org>
References: <20100504071146.GA27482@econcept04.econcept.es>
	<26665FA4-2A27-49EE-A306-3DD19B7B1333@jerris.com>
	<1871ECF8-82DD-4DB6-9C52-C274648CDE4B@freeswitch.org>
Message-ID: <AANLkTimEPiFuKiAw8EwpAqMihpCShUY4uGaugabh6lhp@mail.gmail.com>

On Fri, May 7, 2010 at 2:20 AM, Brian West <brian at freeswitch.org> wrote:

> I just put a patch in for the UTF-8 issue.
> /b

Isaac asked me to thank you, Brian, as the patch is working fine (he
applied the patch to our sources and it's working fine).

-- 
Jaume Sabater
http://linuxsilo.net/

"Ubi sapientas ibi libertas"


From wchao at yahoo.com  Mon May 10 10:09:20 2010
From: wchao at yahoo.com (Wellie Chao)
Date: Mon, 10 May 2010 13:09:20 -0400 (EDT)
Subject: [Freeswitch-users] Voice to text / speech recognition
Message-ID: <alpine.LRH.2.00.1005101227390.20426@osgood.microoffice.com>

I'm looking to provide a feature to transcribe voicemails into text for my 
users. I've looked at SpinVox (which was acquired by Nuance). Their 
licensing method doesn't seem particularly friendly or flexible. They 
charge per user and lock each license to a particular user's actual 
identity. If I have a user named John Smith, I have to buy a license from 
Nuance _for John Smith_. If John Smith stops using the voicemail to text 
feature, but I have another user named Jane Doe who wants to start using 
the voicemail to text feature, I have to buy a new license for Jane Doe. 
Since Nuance makes you pay for a license one year at a time, it gets 
expensive if you have any significant churn. Apart from that, it is just 
annoying to have to register users with another company. I'd like to buy 
100 or 1000 or 10000 licenses and use them how I see fit and not have to 
bother with registering individual users with a provider of speech 
recognition services.

I am wondering if other FreeSWITCH users have recommendations for good 
voice to text services or software. I'd be happy to consider either a 
service or software. If software, ideally it would be free of course, but 
a reasonable cost would also be acceptable. I know about PocketSphinx, but 
is it really sufficiently high quality as to be useful for transcribing 
voicemails? I don't need 100% accuracy -- probably 80% or 90% would be 
good enough since I will also attach the WAV file and direct my users to 
use the WAV file when in doubt.

Any pointers or tips would be appreciated.


From helmut.kuper at ewetel.de  Mon May 10 10:44:03 2010
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Mon, 10 May 2010 19:44:03 +0200
Subject: [Freeswitch-users] bridge_hangup_cause not for internal
	sip	calls?
In-Reply-To: <AANLkTinIevjzNaO3rj9s5vsPKNxTb6iD51rhEeHHcMcy@mail.gmail.com>
References: <4BE7D332.6060302@ewetel.de>
	<AANLkTinIevjzNaO3rj9s5vsPKNxTb6iD51rhEeHHcMcy@mail.gmail.com>
Message-ID: <4BE845E3.9040701@ewetel.de>

Brilliant Mr. Minessale,

you are right. :) So I guess I have to spend some time on refining my 
dialplan, haven't I?

regards
Helmut


Am 10.05.2010 18:08, schrieb Anthony Minessale:
> maybe your TDM is doing early media which would produce a SUCESSFUL 
> originate disposition since the
> origination ends as soon as the call reaches early media.
>



From mike at jerris.com  Mon May 10 11:02:41 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 10 May 2010 14:02:41 -0400
Subject: [Freeswitch-users] signal_bond and uuid
In-Reply-To: <2A08815A59134650A928BAC2C6417D58@MOBILEE1705>
References: <2A08815A59134650A928BAC2C6417D58@MOBILEE1705>
Message-ID: <4A77FD18-26AE-4182-92E1-052109869E95@jerris.com>

The best way to do this is to use rfc2833 dtmf to send the dtmf from the sip client attached to the a leg.

Mike

On May 8, 2010, at 11:25 PM, Madovsky wrote:

> if for example I d like to send dtmf to the legB in api mod_commands
> should I use signal_bond as uuid of the channel ?

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From mike at jerris.com  Mon May 10 11:04:03 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 10 May 2010 14:04:03 -0400
Subject: [Freeswitch-users] default voiceamail sentences
In-Reply-To: <AC78A683BB3B4B6B84755884FD08B792@MOBILEE1705>
References: <AC78A683BB3B4B6B84755884FD08B792@MOBILEE1705>
Message-ID: <99BE8D94-00D8-4828-A326-683658ABDF6C@jerris.com>

This is controlled by the phrase macro's and sound files.  
http://fisheye.freeswitch.org/browse/freeswitch-git/conf/lang/en/vm/sounds.xml?r=HEAD

Mike

On May 9, 2010, at 4:25 AM, Madovsky wrote:

> I'd like to clean the default voicemail answer when user is not registered like
> the person at extension xxxx is not available, record your message after the tone...
> and once the message is registered say "saved, goodbye" only without "press 1 etc.."
> possible ?

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From mike at jerris.com  Mon May 10 11:08:39 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 10 May 2010 14:08:39 -0400
Subject: [Freeswitch-users] luarun fs_cli output
In-Reply-To: <z2o8fa0f3241005091132h7b970f91n8c718fa34bf5cec@mail.gmail.com>
References: <s2p8fa0f3241005080633ue7f5555axf51c8c02cb15da4@mail.gmail.com>
	<8F5F793A-902D-4932-B17A-7E172E039976@gmail.com>
	<z2o8fa0f3241005091132h7b970f91n8c718fa34bf5cec@mail.gmail.com>
Message-ID: <5A3E6BE9-BCA1-4263-8D52-063B2E60D508@jerris.com>

I really must be in person at this location in order to answer this question properly.  Please send plane tickets and hotel information to my email address :D

Mike

On May 9, 2010, at 2:32 PM, Brian Solon wrote:

> Thanks, I'll try that.
> 
> On 9 May 2010 19:05, David Ponzone <david.ponzone at gmail.com> wrote:
> AFAIK, You can't.
> You may send the required output to FS console with the required API call (consoleLog).

> Le 08/05/2010 ? 15:33, Brian Solon a ?crit :
> 
>> Hi, 
>> 
>> This is my first post here so I just wanted to say FreeSWITCH is fantastic. I'm using it for an art installation which has been up and running for the last week in a gallery in Sligo, Ireland. ( More info here, for the curious: http://facebook.com/HelloOperatorIE ).
>> 
>> Just one question for now: how can I see the output of a Lua script when launched from fs_cli?
>> 
>> -bash-3.2$ pwd
>> /opt/freeswitch/scripts
>> -bash-3.2$ lua
>> Lua 5.1.2  Copyright (C) 1994-2007 Lua.org, PUC-Rio
>> > require "test"
>> Hello World
>> >
>> 
>> I'm probably missing something simple, but when I run it from the CLI all I get is "+OK":
>> 
>> freeswitch at internal> luarun test.lua
>> +OK

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From mike at jerris.com  Mon May 10 11:16:36 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 10 May 2010 14:16:36 -0400
Subject: [Freeswitch-users] listening on multiple events in mod_managed
In-Reply-To: <AANLkTik0oTbEHStsteb0n4RX8bian_CG_r-5qJxKLYN1@mail.gmail.com>
References: <AANLkTik0oTbEHStsteb0n4RX8bian_CG_r-5qJxKLYN1@mail.gmail.com>
Message-ID: <07E0AE25-6AB4-4754-B775-10E8EE1816E9@jerris.com>

It appears that an event consumer can only be 1 event.  You can do multiple consumers or could extend EventConsumer to support multiple event bindings.

Mike

On May 10, 2010, at 1:28 AM, babak yakhchali wrote:

> Hi
> How is it possible to listen for maore than one type of events but not all of them something like
> EventConsumer con = new EventConsumer("CUSTOM CHANNEL_DESTROY CHANNEL_HANGUP", null);



From anthony.minessale at gmail.com  Mon May 10 11:25:57 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 10 May 2010 13:25:57 -0500
Subject: [Freeswitch-users] listening on multiple events in mod_managed
In-Reply-To: <AANLkTik0oTbEHStsteb0n4RX8bian_CG_r-5qJxKLYN1@mail.gmail.com>
References: <AANLkTik0oTbEHStsteb0n4RX8bian_CG_r-5qJxKLYN1@mail.gmail.com>
Message-ID: <AANLkTikIUMjZGZtNMxtnxd1gAU8KMHzSIgkAoOMG_QMZ@mail.gmail.com>

This was a reasonable request, so I added a patch to HEAD to allow a new
bind() method.
This takes the same arguments as the constructor and can be called multiple
times on a constructed event.
In addition the constructor can now be called with no arguments, followed by
many bind calls.



On Mon, May 10, 2010 at 12:28 AM, babak yakhchali <
babak.freeswitch at gmail.com> wrote:

> Hi
> How is it possible to listen for maore than one type of events but not all
> of them something like
> EventConsumer con = new EventConsumer("CUSTOM CHANNEL_DESTROY
> CHANNEL_HANGUP", null);
> thanks
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
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
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From infos at madovsky.org  Mon May 10 12:13:38 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 10 May 2010 15:13:38 -0400
Subject: [Freeswitch-users] default voiceamail sentences
References: <AC78A683BB3B4B6B84755884FD08B792@MOBILEE1705>
	<99BE8D94-00D8-4828-A326-683658ABDF6C@jerris.com>
Message-ID: <4A702286E1DB4DC6B81D7405E1991505@MOBILEE1705>

Thx Mike,
but now I get this error on log console:

2010-05-10 15:12:32.256310 [ERR] switch_ivr_play_say.c:150 Can't find macro voicemail_record_file_check.
2010-05-10 15:12:32.256310 [WARNING] switch_ivr_play_say.c:339 Macro [voicemail_record_file_check]: '1:2:3' did not match any patterns
2010-05-10 15:12:32.956370 [ERR] switch_ivr_play_say.c:150 Can't find macro voicemail_record_urgent_check.
2010-05-10 15:12:32.956370 [WARNING] switch_ivr_play_say.c:339 Macro [voicemail_record_urgent_check]: '*:#' did not match any patterns

is there another files where the voicemail is managed ?

Thanks

F
  ----- Original Message ----- 
  From: Michael Jerris 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, May 10, 2010 2:04 PM
  Subject: Re: [Freeswitch-users] default voiceamail sentences


  This is controlled by the phrase macro's and sound files.  
  http://fisheye.freeswitch.org/browse/freeswitch-git/conf/lang/en/vm/sounds.xml?r=HEAD


  Mike


  On May 9, 2010, at 4:25 AM, Madovsky wrote:


    I'd like to clean the default voicemail answer when user is not registered like
    the person at extension xxxx is not available, record your message after the tone...
    and once the message is registered say "saved, goodbye" only without "press 1 etc.."
    possible ?




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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From anthony.minessale at gmail.com  Mon May 10 12:21:07 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 10 May 2010 14:21:07 -0500
Subject: [Freeswitch-users] default voiceamail sentences
In-Reply-To: <4A702286E1DB4DC6B81D7405E1991505@MOBILEE1705>
References: <AC78A683BB3B4B6B84755884FD08B792@MOBILEE1705>
	<99BE8D94-00D8-4828-A326-683658ABDF6C@jerris.com>
	<4A702286E1DB4DC6B81D7405E1991505@MOBILEE1705>
Message-ID: <AANLkTindTHe5PIDBxXFgTwf6wuFFZFU7Bq5qj3Ce1yq3@mail.gmail.com>

you seemingly broke the vm template.
do make vm-sync to restore it.


On Mon, May 10, 2010 at 2:13 PM, Madovsky <infos at madovsky.org> wrote:

>  Thx Mike,
> but now I get this error on log console:
>
> 2010-05-10 15:12:32.256310 [ERR] switch_ivr_play_say.c:150 Can't find macro
> voicemail_record_file_check.
> 2010-05-10 15:12:32.256310 [WARNING] switch_ivr_play_say.c:339 Macro
> [voicemail_record_file_check]: '1:2:3' did not match any patterns
> 2010-05-10 15:12:32.956370 [ERR] switch_ivr_play_say.c:150 Can't find macro
> voicemail_record_urgent_check.
> 2010-05-10 15:12:32.956370 [WARNING] switch_ivr_play_say.c:339 Macro
> [voicemail_record_urgent_check]: '*:#' did not match any patterns
> is there another files where the voicemail is managed ?
>
> Thanks
>
> F
>
> ----- Original Message -----
> *From:* Michael Jerris <mike at jerris.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Monday, May 10, 2010 2:04 PM
> *Subject:* Re: [Freeswitch-users] default voiceamail sentences
>
> This is controlled by the phrase macro's and sound files.
>
> http://fisheye.freeswitch.org/browse/freeswitch-git/conf/lang/en/vm/sounds.xml?r=HEAD
>
> Mike
>
>  On May 9, 2010, at 4:25 AM, Madovsky wrote:
>
>  I'd like to clean the default voicemail answer when user is not
> registered like
> the person at extension xxxx is not available, record your message after
> the tone...
> and once the message is registered say "saved, goodbye" only without "press
> 1 etc.."
> possible ?
>
>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
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
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From infos at madovsky.org  Mon May 10 12:28:14 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 10 May 2010 15:28:14 -0400
Subject: [Freeswitch-users] default voiceamail sentences
References: <AC78A683BB3B4B6B84755884FD08B792@MOBILEE1705><99BE8D94-00D8-4828-A326-683658ABDF6C@jerris.com><4A702286E1DB4DC6B81D7405E1991505@MOBILEE1705>
	<AANLkTindTHe5PIDBxXFgTwf6wuFFZFU7Bq5qj3Ce1yq3@mail.gmail.com>
Message-ID: <352AE9C7D4BC4B7398A699DB016F6F76@MOBILEE1705>

ok but I dont' want also these option, I d like to remove them

Thanks

F
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, May 10, 2010 3:21 PM
  Subject: Re: [Freeswitch-users] default voiceamail sentences


  you seemingly broke the vm template.
  do make vm-sync to restore it.



  On Mon, May 10, 2010 at 2:13 PM, Madovsky <infos at madovsky.org> wrote:

    Thx Mike,
    but now I get this error on log console:

    2010-05-10 15:12:32.256310 [ERR] switch_ivr_play_say.c:150 Can't find macro voicemail_record_file_check.
    2010-05-10 15:12:32.256310 [WARNING] switch_ivr_play_say.c:339 Macro [voicemail_record_file_check]: '1:2:3' did not match any patterns
    2010-05-10 15:12:32.956370 [ERR] switch_ivr_play_say.c:150 Can't find macro voicemail_record_urgent_check.
    2010-05-10 15:12:32.956370 [WARNING] switch_ivr_play_say.c:339 Macro [voicemail_record_urgent_check]: '*:#' did not match any patterns

    is there another files where the voicemail is managed ?

    Thanks

    F
      ----- Original Message ----- 
      From: Michael Jerris 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Monday, May 10, 2010 2:04 PM
      Subject: Re: [Freeswitch-users] default voiceamail sentences


      This is controlled by the phrase macro's and sound files.   
      http://fisheye.freeswitch.org/browse/freeswitch-git/conf/lang/en/vm/sounds.xml?r=HEAD


      Mike


      On May 9, 2010, at 4:25 AM, Madovsky wrote:


        I'd like to clean the default voicemail answer when user is not registered like
        the person at extension xxxx is not available, record your message after the tone...
        and once the message is registered say "saved, goodbye" only without "press 1 etc.."
        possible ?




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


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



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





  -- 
  Anthony Minessale II

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

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

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



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


  _______________________________________________
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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From anthony.minessale at gmail.com  Mon May 10 12:37:27 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 10 May 2010 14:37:27 -0500
Subject: [Freeswitch-users] default voiceamail sentences
In-Reply-To: <352AE9C7D4BC4B7398A699DB016F6F76@MOBILEE1705>
References: <AC78A683BB3B4B6B84755884FD08B792@MOBILEE1705>
	<99BE8D94-00D8-4828-A326-683658ABDF6C@jerris.com>
	<4A702286E1DB4DC6B81D7405E1991505@MOBILEE1705>
	<AANLkTindTHe5PIDBxXFgTwf6wuFFZFU7Bq5qj3Ce1yq3@mail.gmail.com>
	<352AE9C7D4BC4B7398A699DB016F6F76@MOBILEE1705>
Message-ID: <AANLkTilRpsk5xxDsh8Yee6tedADH9hnmv4l9D1RZglky@mail.gmail.com>

don't remove the entire macro then just the actions inside them

On Mon, May 10, 2010 at 2:28 PM, Madovsky <infos at madovsky.org> wrote:

>  ok but I dont' want also these option, I d like to remove them
>
> Thanks
>
> F
>
> ----- Original Message -----
> *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Monday, May 10, 2010 3:21 PM
> *Subject:* Re: [Freeswitch-users] default voiceamail sentences
>
> you seemingly broke the vm template.
> do make vm-sync to restore it.
>
>
> On Mon, May 10, 2010 at 2:13 PM, Madovsky <infos at madovsky.org> wrote:
>
>>  Thx Mike,
>> but now I get this error on log console:
>>
>> 2010-05-10 15:12:32.256310 [ERR] switch_ivr_play_say.c:150 Can't find
>> macro voicemail_record_file_check.
>> 2010-05-10 15:12:32.256310 [WARNING] switch_ivr_play_say.c:339 Macro
>> [voicemail_record_file_check]: '1:2:3' did not match any patterns
>> 2010-05-10 15:12:32.956370 [ERR] switch_ivr_play_say.c:150 Can't find
>> macro voicemail_record_urgent_check.
>> 2010-05-10 15:12:32.956370 [WARNING] switch_ivr_play_say.c:339 Macro
>> [voicemail_record_urgent_check]: '*:#' did not match any patterns
>> is there another files where the voicemail is managed ?
>>
>> Thanks
>>
>> F
>>
>>   ----- Original Message -----
>> *From:* Michael Jerris <mike at jerris.com>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Sent:* Monday, May 10, 2010 2:04 PM
>> *Subject:* Re: [Freeswitch-users] default voiceamail sentences
>>
>> This is controlled by the phrase macro's and sound files.
>>
>> http://fisheye.freeswitch.org/browse/freeswitch-git/conf/lang/en/vm/sounds.xml?r=HEAD
>>
>> Mike
>>
>>  On May 9, 2010, at 4:25 AM, Madovsky wrote:
>>
>>  I'd like to clean the default voicemail answer when user is not
>> registered like
>> the person at extension xxxx is not available, record your message after
>> the tone...
>> and once the message is registered say "saved, goodbye" only without
>> "press 1 etc.."
>> possible ?
>>
>>
>>  ------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
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>
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From infos at madovsky.org  Mon May 10 12:54:28 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 10 May 2010 15:54:28 -0400
Subject: [Freeswitch-users] default voiceamail sentences
References: <AC78A683BB3B4B6B84755884FD08B792@MOBILEE1705><99BE8D94-00D8-4828-A326-683658ABDF6C@jerris.com><4A702286E1DB4DC6B81D7405E1991505@MOBILEE1705><AANLkTindTHe5PIDBxXFgTwf6wuFFZFU7Bq5qj3Ce1yq3@mail.gmail.com><352AE9C7D4BC4B7398A699DB016F6F76@MOBILEE1705>
	<AANLkTilRpsk5xxDsh8Yee6tedADH9hnmv4l9D1RZglky@mail.gmail.com>
Message-ID: <DD280598F71D49AD8054CF2F8CEBD5BA@MOBILEE1705>

Ok it's what I'm trying to do.. but at the end the girl says 2 times message saved and half of goodbye.. ;)
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, May 10, 2010 3:37 PM
  Subject: Re: [Freeswitch-users] default voiceamail sentences


  don't remove the entire macro then just the actions inside them


  On Mon, May 10, 2010 at 2:28 PM, Madovsky <infos at madovsky.org> wrote:

    ok but I dont' want also these option, I d like to remove them

    Thanks

    F
      ----- Original Message ----- 
      From: Anthony Minessale 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Monday, May 10, 2010 3:21 PM
      Subject: Re: [Freeswitch-users] default voiceamail sentences


      you seemingly broke the vm template. 
      do make vm-sync to restore it.



      On Mon, May 10, 2010 at 2:13 PM, Madovsky <infos at madovsky.org> wrote:

        Thx Mike,
        but now I get this error on log console:

        2010-05-10 15:12:32.256310 [ERR] switch_ivr_play_say.c:150 Can't find macro voicemail_record_file_check.
        2010-05-10 15:12:32.256310 [WARNING] switch_ivr_play_say.c:339 Macro [voicemail_record_file_check]: '1:2:3' did not match any patterns
        2010-05-10 15:12:32.956370 [ERR] switch_ivr_play_say.c:150 Can't find macro voicemail_record_urgent_check.
        2010-05-10 15:12:32.956370 [WARNING] switch_ivr_play_say.c:339 Macro [voicemail_record_urgent_check]: '*:#' did not match any patterns

        is there another files where the voicemail is managed ?

        Thanks

        F
          ----- Original Message ----- 
          From: Michael Jerris 
          To: freeswitch-users at lists.freeswitch.org 
          Sent: Monday, May 10, 2010 2:04 PM
          Subject: Re: [Freeswitch-users] default voiceamail sentences


          This is controlled by the phrase macro's and sound files.   
          http://fisheye.freeswitch.org/browse/freeswitch-git/conf/lang/en/vm/sounds.xml?r=HEAD


          Mike


          On May 9, 2010, at 4:25 AM, Madovsky wrote:


            I'd like to clean the default voicemail answer when user is not registered like
            the person at extension xxxx is not available, record your message after the tone...
            and once the message is registered say "saved, goodbye" only without "press 1 etc.."
            possible ?




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


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



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





      -- 
      Anthony Minessale II

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



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





  -- 
  Anthony Minessale II

  FreeSWITCH http://www.freeswitch.org/
  ClueCon http://www.cluecon.com/
  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-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From neilp at cs.stanford.edu  Mon May 10 14:03:56 2010
From: neilp at cs.stanford.edu (Neil Patel)
Date: Mon, 10 May 2010 14:03:56 -0700
Subject: [Freeswitch-users] how do I detect a call has rung once?
Message-ID: <u2je5503bdb1005101403u405a5eebo9a2b8c698e9cffaf@mail.gmail.com>

I want to initiate a missed call from FS (via lua script). Is there anything
in the early media I can check to see that the call has rung at the endpoint
at least once?

Thanks,
Neil
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From anthony.minessale at gmail.com  Mon May 10 14:27:07 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 10 May 2010 16:27:07 -0500
Subject: [Freeswitch-users] how do I detect a call has rung once?
In-Reply-To: <u2je5503bdb1005101403u405a5eebo9a2b8c698e9cffaf@mail.gmail.com>
References: <u2je5503bdb1005101403u405a5eebo9a2b8c698e9cffaf@mail.gmail.com>
Message-ID: <AANLkTinS6No0-1Z1oe9qF6T_9yVedPl3JI-T2OopEoep@mail.gmail.com>

as soon as you get the first early media signal, that is how you know.


On Mon, May 10, 2010 at 4:03 PM, Neil Patel <neilp at cs.stanford.edu> wrote:

> I want to initiate a missed call from FS (via lua script). Is there
> anything in the early media I can check to see that the call has rung at the
> endpoint at least once?
>
> Thanks,
> Neil
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <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
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From sos at sokhapkin.dyndns.org  Mon May 10 14:34:09 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 10 May 2010 17:34:09 -0400
Subject: [Freeswitch-users] how do I detect a call has rung once?
In-Reply-To: <AANLkTinS6No0-1Z1oe9qF6T_9yVedPl3JI-T2OopEoep@mail.gmail.com>
References: <u2je5503bdb1005101403u405a5eebo9a2b8c698e9cffaf@mail.gmail.com>
	<AANLkTinS6No0-1Z1oe9qF6T_9yVedPl3JI-T2OopEoep@mail.gmail.com>
Message-ID: <201005101734.09903.sos@sokhapkin.dyndns.org>

Unfortunately this is a wrong assumption. You can get dead air in early media 
and SIP error after that.

On Monday 10 May 2010, Anthony Minessale wrote:
> as soon as you get the first early media signal, that is how you know.
> 
> On Mon, May 10, 2010 at 4:03 PM, Neil Patel <neilp at cs.stanford.edu> wrote:
> > I want to initiate a missed call from FS (via lua script). Is there
> > anything in the early media I can check to see that the call has rung at
> > the endpoint at least once?
> >
> > Thanks,
> > Neil
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 



From krice at freeswitch.org  Mon May 10 14:40:50 2010
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 10 May 2010 16:40:50 -0500
Subject: [Freeswitch-users] how do I detect a call has rung once?
In-Reply-To: <201005101734.09903.sos@sokhapkin.dyndns.org>
Message-ID: <C80DE792.6CF3D%krice@freeswitch.org>

Unfortunately you can never guarentee that the far end has range atleast
once...

Session progress (on any network topology sip, h323, ss7 or even PRI) does
not guarantee the far end has rang only that the switch you sent the call to
has indicated that it is attempting to ring the far end...

Then on top of that add any number of different user devices (softphones,
hardphones, analog cordless phones whatever) who knows when they'll actually
start ringing... I have had digital cordless phones that would never ring on
the first ring pulse on an analog line while the old school 500 set next to
them was just ringing away...

All you can do is take your best guess at it... ie: wait 4 or 5 seconds
after session progress then hope that was 1 ring due to typical ring cycle
length

K


On 5/10/10 4:34 PM, "Sergey Okhapkin" <sos at sokhapkin.dyndns.org> wrote:

> Unfortunately this is a wrong assumption. You can get dead air in early media
> and SIP error after that.
> 
> On Monday 10 May 2010, Anthony Minessale wrote:
>> as soon as you get the first early media signal, that is how you know.
>> 
>> On Mon, May 10, 2010 at 4:03 PM, Neil Patel <neilp at cs.stanford.edu> wrote:
>>> I want to initiate a missed call from FS (via lua script). Is there
>>> anything in the early media I can check to see that the call has rung at
>>> the endpoint at least once?
>>> 
>>> Thanks,
>>> Neil
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From christian.loeschenkohl at xpirio.com  Mon May 10 14:45:44 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Mon, 10 May 2010 23:45:44 +0200
Subject: [Freeswitch-users] blind transfer and outbound socket
Message-ID: <4BE87E88.2000106@xpirio.com>

hello

i have a problem with a freeswitch hosted pbx system.
when i call in to a extension (e.g. 40) an answer the call and then do a blind transfer to extension 50 it
doesn't work.
the outbound socket script do not work or do not get called in this case (no mail from the script).

on the other hand if i do a attended transfer it works as it should (sending mail in the script works).
the same context, the same scripts.

we use snom 320 phones, the sip refer looks normal.
blind transfer also doesn't work with a softclient (sjphone)

where could i start?

br

-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com


From anthony.minessale at gmail.com  Mon May 10 14:49:06 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 10 May 2010 16:49:06 -0500
Subject: [Freeswitch-users] how do I detect a call has rung once?
In-Reply-To: <201005101734.09903.sos@sokhapkin.dyndns.org>
References: <u2je5503bdb1005101403u405a5eebo9a2b8c698e9cffaf@mail.gmail.com>
	<AANLkTinS6No0-1Z1oe9qF6T_9yVedPl3JI-T2OopEoep@mail.gmail.com>
	<201005101734.09903.sos@sokhapkin.dyndns.org>
Message-ID: <AANLkTilVHAdZDgF3a5Mg2HgXrXOg6hkrLKGb9xUAAdik@mail.gmail.com>

Unfortunately it's not an assumption.

You have to consider that things are sane.  If they are.  When you place a
call and you get either a 180 or 183 it denotes that ringing has been
indicated by the far end.  If you want to get into the gory details they map
over to a very specific set of ISDN signals that indicate that the phone has
rung or not both with and without media.  In either case both with and
without audio if FreeSWITCH gets the early media indication it means someone
has told us there is ringing.

If things are not sane, you cannot do anything but find another route.





On Mon, May 10, 2010 at 4:34 PM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> Unfortunately this is a wrong assumption. You can get dead air in early
> media
> and SIP error after that.
>
> On Monday 10 May 2010, Anthony Minessale wrote:
> > as soon as you get the first early media signal, that is how you know.
> >
> > On Mon, May 10, 2010 at 4:03 PM, Neil Patel <neilp at cs.stanford.edu>
> wrote:
> > > I want to initiate a missed call from FS (via lua script). Is there
> > > anything in the early media I can check to see that the call has rung
> at
> > > the endpoint at least once?
> > >
> > > Thanks,
> > > Neil
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
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
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From anthony.minessale at gmail.com  Mon May 10 14:54:02 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 10 May 2010 16:54:02 -0500
Subject: [Freeswitch-users] blind transfer and outbound socket
In-Reply-To: <4BE87E88.2000106@xpirio.com>
References: <4BE87E88.2000106@xpirio.com>
Message-ID: <AANLkTikcfzQS6uuSHiCAdllO5ApvAiUT_l1aXMSU3HLT@mail.gmail.com>

what exactly does your script call to try to make it transfer?

2010/5/10 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>

> hello
>
> i have a problem with a freeswitch hosted pbx system.
> when i call in to a extension (e.g. 40) an answer the call and then do a
> blind transfer to extension 50 it
> doesn't work.
> the outbound socket script do not work or do not get called in this case
> (no mail from the script).
>
> on the other hand if i do a attended transfer it works as it should
> (sending mail in the script works).
> the same context, the same scripts.
>
> we use snom 320 phones, the sip refer looks normal.
> blind transfer also doesn't work with a softclient (sjphone)
>
> where could i start?
>
> br
>
> --
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
>
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
>
> T  +43 (0) 5 77 11 - 1000
> F  +43 (0) 5 77 11 - 1002
> E  christian.loeschenkohl at xpirio.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <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
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From christian.loeschenkohl at xpirio.com  Mon May 10 15:13:41 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Tue, 11 May 2010 00:13:41 +0200
Subject: [Freeswitch-users] blind transfer and outbound socket
In-Reply-To: <AANLkTikcfzQS6uuSHiCAdllO5ApvAiUT_l1aXMSU3HLT@mail.gmail.com>
References: <4BE87E88.2000106@xpirio.com>
	<AANLkTikcfzQS6uuSHiCAdllO5ApvAiUT_l1aXMSU3HLT@mail.gmail.com>
Message-ID: <4BE88515.30105@xpirio.com>

the scripts do some database lookups (setting callers name) and do find the called
user e.g. expand the called 50 to 50 at customer.domain.com and finaly do the call bridge.

the transfer itself is done by the phone/softclient with a refer packet, as far as i can see.

a debug trace is here
http://pastebin.freeswitch.org/12958

br

Anthony Minessale wrote:

> what exactly does your script call to try to make it transfer?
> 
> 2010/5/10 Christian L?schenkohl <christian.loeschenkohl at xpirio.com 
> <mailto:christian.loeschenkohl at xpirio.com>>
> 
>     hello
> 
>     i have a problem with a freeswitch hosted pbx system.
>     when i call in to a extension (e.g. 40) an answer the call and then
>     do a blind transfer to extension 50 it
>     doesn't work.
>     the outbound socket script do not work or do not get called in this
>     case (no mail from the script).
> 
>     on the other hand if i do a attended transfer it works as it should
>     (sending mail in the script works).
>     the same context, the same scripts.
> 
>     we use snom 320 phones, the sip refer looks normal.
>     blind transfer also doesn't work with a softclient (sjphone)
> 
>     where could i start?
> 
>     br
> 
>     --
>     Ing. Christian L?schenkohl
>     Technische Leitung, Forschung & Entwicklung VoIP
> 
>     xpirio
>     Telekommunikation & Service GmbH
>     Lakeside B04
>     9020 Klagenfurt
>     Austria
> 
>     T  +43 (0) 5 77 11 - 1000
>     F  +43 (0) 5 77 11 - 1002
>     E  christian.loeschenkohl at xpirio.com
>     <mailto:christian.loeschenkohl at xpirio.com>
> 
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com


From anthony.minessale at gmail.com  Mon May 10 15:23:17 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 10 May 2010 17:23:17 -0500
Subject: [Freeswitch-users] blind transfer and outbound socket
In-Reply-To: <4BE88515.30105@xpirio.com>
References: <4BE87E88.2000106@xpirio.com>
	<AANLkTikcfzQS6uuSHiCAdllO5ApvAiUT_l1aXMSU3HLT@mail.gmail.com>
	<4BE88515.30105@xpirio.com>
Message-ID: <AANLkTinftpVgycyYVPLh6-xS9aLPFnReMz3AjAVtIfDh@mail.gmail.com>

it looks like it works fine to me.

the call transfers back to the dialplan where it again executes your socket
app and that app either does nothing
or hangs up.

See line 789 of your pastebin.


2010/5/10 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>

> the scripts do some database lookups (setting callers name) and do find the
> called
> user e.g. expand the called 50 to 50 at customer.domain.com and finaly do the
> call bridge.
>
> the transfer itself is done by the phone/softclient with a refer packet, as
> far as i can see.
>
> a debug trace is here
> http://pastebin.freeswitch.org/12958
>
> br
>
> Anthony Minessale wrote:
>
> > what exactly does your script call to try to make it transfer?
> >
> > 2010/5/10 Christian L?schenkohl <christian.loeschenkohl at xpirio.com
> > <mailto:christian.loeschenkohl at xpirio.com>>
> >
> >     hello
> >
> >     i have a problem with a freeswitch hosted pbx system.
> >     when i call in to a extension (e.g. 40) an answer the call and then
> >     do a blind transfer to extension 50 it
> >     doesn't work.
> >     the outbound socket script do not work or do not get called in this
> >     case (no mail from the script).
> >
> >     on the other hand if i do a attended transfer it works as it should
> >     (sending mail in the script works).
> >     the same context, the same scripts.
> >
> >     we use snom 320 phones, the sip refer looks normal.
> >     blind transfer also doesn't work with a softclient (sjphone)
> >
> >     where could i start?
> >
> >     br
> >
> >     --
> >     Ing. Christian L?schenkohl
> >     Technische Leitung, Forschung & Entwicklung VoIP
> >
> >     xpirio
> >     Telekommunikation & Service GmbH
> >     Lakeside B04
> >     9020 Klagenfurt
> >     Austria
> >
> >     T  +43 (0) 5 77 11 - 1000
> >     F  +43 (0) 5 77 11 - 1002
> >     E  christian.loeschenkohl at xpirio.com
> >     <mailto:christian.loeschenkohl at xpirio.com>
> >
> >     _______________________________________________
> >     FreeSWITCH-users mailing list
> >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     http://www.freeswitch.org
> >
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> > IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> > pstn:+19193869900
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> --
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
>
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
>
> T  +43 (0) 5 77 11 - 1000
> F  +43 (0) 5 77 11 - 1002
> E  christian.loeschenkohl at xpirio.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <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
-------------- next part --------------
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From christian.loeschenkohl at xpirio.com  Mon May 10 15:45:44 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Tue, 11 May 2010 00:45:44 +0200
Subject: [Freeswitch-users] blind transfer and outbound socket
In-Reply-To: <AANLkTinftpVgycyYVPLh6-xS9aLPFnReMz3AjAVtIfDh@mail.gmail.com>
References: <4BE87E88.2000106@xpirio.com>	<AANLkTikcfzQS6uuSHiCAdllO5ApvAiUT_l1aXMSU3HLT@mail.gmail.com>	<4BE88515.30105@xpirio.com>
	<AANLkTinftpVgycyYVPLh6-xS9aLPFnReMz3AjAVtIfDh@mail.gmail.com>
Message-ID: <4BE88C98.5050107@xpirio.com>

yes, and this is my problem
it looks normal but the scripts are not executed (i send a email in the fist line of
the script and there is no mail send to me here).
the call is hung up without much comment, the dialplan runs through until the hangup -
without the bridge command executed from the script.

attended transfer is here, with this it works
http://pastebin.freeswitch.org/12959

the scripts do also work if i call the extension directly.
the extension 50 also rings if i put the bridge in the dialplan - my only problem here
is that it looks like the script is called but it is never executed.

br



Anthony Minessale wrote:

> it looks like it works fine to me.
> 
> the call transfers back to the dialplan where it again executes your 
> socket app and that app either does nothing
> or hangs up.
> 
> See line 789 of your pastebin.
> 
> 
> 2010/5/10 Christian L?schenkohl <christian.loeschenkohl at xpirio.com 
> <mailto:christian.loeschenkohl at xpirio.com>>
> 
>     the scripts do some database lookups (setting callers name) and do
>     find the called
>     user e.g. expand the called 50 to 50 at customer.domain.com
>     <mailto:50 at customer.domain.com> and finaly do the call bridge.
> 
>     the transfer itself is done by the phone/softclient with a refer
>     packet, as far as i can see.
> 
>     a debug trace is here
>     http://pastebin.freeswitch.org/12958
> 
>     br
> 
>     Anthony Minessale wrote:
> 
>      > what exactly does your script call to try to make it transfer?
>      >
>      > 2010/5/10 Christian L?schenkohl
>     <christian.loeschenkohl at xpirio.com
>     <mailto:christian.loeschenkohl at xpirio.com>
>      > <mailto:christian.loeschenkohl at xpirio.com
>     <mailto:christian.loeschenkohl at xpirio.com>>>
>      >
>      >     hello
>      >
>      >     i have a problem with a freeswitch hosted pbx system.
>      >     when i call in to a extension (e.g. 40) an answer the call
>     and then
>      >     do a blind transfer to extension 50 it
>      >     doesn't work.
>      >     the outbound socket script do not work or do not get called
>     in this
>      >     case (no mail from the script).
>      >
>      >     on the other hand if i do a attended transfer it works as it
>     should
>      >     (sending mail in the script works).
>      >     the same context, the same scripts.
>      >
>      >     we use snom 320 phones, the sip refer looks normal.
>      >     blind transfer also doesn't work with a softclient (sjphone)
>      >
>      >     where could i start?
>      >
>      >     br
>      >
>      >     --
>      >     Ing. Christian L?schenkohl
>      >     Technische Leitung, Forschung & Entwicklung VoIP
>      >
>      >     xpirio
>      >     Telekommunikation & Service GmbH
>      >     Lakeside B04
>      >     9020 Klagenfurt
>      >     Austria
>      >
>      >     T  +43 (0) 5 77 11 - 1000
>      >     F  +43 (0) 5 77 11 - 1002
>      >     E  christian.loeschenkohl at xpirio.com
>     <mailto:christian.loeschenkohl at xpirio.com>
>      >     <mailto:christian.loeschenkohl at xpirio.com
>     <mailto:christian.loeschenkohl at xpirio.com>>
>      >
>      >     _______________________________________________
>      >     FreeSWITCH-users mailing list
>      >     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>      >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>      >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>      >    
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>      >     http://www.freeswitch.org
>      >
>      >
>      >
>      >
>      > --
>      > Anthony Minessale II
>      >
>      > FreeSWITCH http://www.freeswitch.org/
>      > ClueCon http://www.cluecon.com/
>      > Twitter: http://twitter.com/FreeSWITCH_wire
>      >
>      > AIM: anthm
>      > MSN:anthony_minessale at hotmail.com
>     <mailto:MSN%3Aanthony_minessale at hotmail.com>
>      > <mailto:MSN%3Aanthony_minessale at hotmail.com
>     <mailto:MSN%253Aanthony_minessale at hotmail.com>>
>      > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>     <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>      > <mailto:PAYPAL%3Aanthony.minessale at gmail.com
>     <mailto:PAYPAL%253Aanthony.minessale at gmail.com>>
>      > IRC: irc.freenode.net <http://irc.freenode.net>
>     <http://irc.freenode.net> #freeswitch
>      >
>      > FreeSWITCH Developer Conference
>      > sip:888 at conference.freeswitch.org
>     <mailto:sip%3A888 at conference.freeswitch.org>
>      > <mailto:sip%3A888 at conference.freeswitch.org
>     <mailto:sip%253A888 at conference.freeswitch.org>>
>      > googletalk:conf+888 at conference.freeswitch.org
>     <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>      > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org
>     <mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org>>
>      > pstn:+19193869900
>      >
>      >
>      >
>     ------------------------------------------------------------------------
>      >
>      > _______________________________________________
>      > FreeSWITCH-users mailing list
>      > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>      > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>      >
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>      > http://www.freeswitch.org
> 
> 
>     --
>     Ing. Christian L?schenkohl
>     Technische Leitung, Forschung & Entwicklung VoIP
> 
>     xpirio
>     Telekommunikation & Service GmbH
>     Lakeside B04
>     9020 Klagenfurt
>     Austria
> 
>     T  +43 (0) 5 77 11 - 1000
>     F  +43 (0) 5 77 11 - 1002
>     E  christian.loeschenkohl at xpirio.com
>     <mailto:christian.loeschenkohl at xpirio.com>
> 
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com


From jan.berger at video24.no  Mon May 10 15:49:41 2010
From: jan.berger at video24.no (Jan Berger)
Date: Tue, 11 May 2010 00:49:41 +0200
Subject: [Freeswitch-users] Voice to text / speech recognition
In-Reply-To: <alpine.LRH.2.00.1005101227390.20426@osgood.microoffice.com>
References: <alpine.LRH.2.00.1005101227390.20426@osgood.microoffice.com>
Message-ID: <09FF0B4B64784D0AA874349E55F2C0C9@dell9400>

Hi,

I am going to look into Sphinx myself for IVR menus. 

Recognition factor is not so much the software as it is the quality of the
database and how well it is tuned. Tuning you can do yourself with tools
provided from sphinx + it does exist a separate open source database that I
believe you can use. 

Doing IVR menus is feasible. But, to translate free speech you have a
challenge called dialects. The fact is that recognition factor is not 100%
even with people with the same dialect, and many dialects are very far out
so analyze the language of your clients and make some very accurate
questions to the vendor to ensure you get what you want.

Jan

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Wellie
Chao
Sent: 10. mai 2010 19:09
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Voice to text / speech recognition

I'm looking to provide a feature to transcribe voicemails into text for my 
users. I've looked at SpinVox (which was acquired by Nuance). Their 
licensing method doesn't seem particularly friendly or flexible. They 
charge per user and lock each license to a particular user's actual 
identity. If I have a user named John Smith, I have to buy a license from 
Nuance _for John Smith_. If John Smith stops using the voicemail to text 
feature, but I have another user named Jane Doe who wants to start using 
the voicemail to text feature, I have to buy a new license for Jane Doe. 
Since Nuance makes you pay for a license one year at a time, it gets 
expensive if you have any significant churn. Apart from that, it is just 
annoying to have to register users with another company. I'd like to buy 
100 or 1000 or 10000 licenses and use them how I see fit and not have to 
bother with registering individual users with a provider of speech 
recognition services.

I am wondering if other FreeSWITCH users have recommendations for good 
voice to text services or software. I'd be happy to consider either a 
service or software. If software, ideally it would be free of course, but 
a reasonable cost would also be acceptable. I know about PocketSphinx, but 
is it really sufficiently high quality as to be useful for transcribing 
voicemails? I don't need 100% accuracy -- probably 80% or 90% would be 
good enough since I will also attach the WAV file and direct my users to 
use the WAV file when in doubt.

Any pointers or tips would be appreciated.

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



From msc at freeswitch.org  Mon May 10 15:57:43 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 10 May 2010 15:57:43 -0700
Subject: [Freeswitch-users] outgoing gateway
In-Reply-To: <1419115537-1273493954-cardhu_decombobulator_blackberry.rim.net-840701352-@bda057.bisx.prodap.on.blackberry>
References: <472BC89E-11F0-49C2-8EA5-23783BC38770@jerris.com>
	<y2z23f91031005071930p3f96278aj9f7db3a5e0df60aa@mail.gmail.com>
	<r2h547ec34a1005080126i9e6f0cche81e1c1118ece6ef@mail.gmail.com>
	<09D47C25-81C0-4815-B477-7595D769EF60@gmail.com>
	<h2s547ec34a1005081716xc09243cq20ec07fda9ba863c@mail.gmail.com>
	<0EC75EB8-42A9-4C56-9DA7-27C8AA642131@gmail.com>
	<v2o547ec34a1005091506m991f371aj9213eb5e4965ea80@mail.gmail.com>
	<o2y547ec34a1005091542p219ff4c6w5b2d64041f511e85@mail.gmail.com>
	<20100510001816.GA4318@jdc.jasonjgw.net>
	<1419115537-1273493954-cardhu_decombobulator_blackberry.rim.net-840701352-@bda057.bisx.prodap.on.blackberry>
Message-ID: <AANLkTikSvL5MA5omxcbe8rZ_Js6Me0h8LfXyMhQWvQT-@mail.gmail.com>

On Mon, May 10, 2010 at 5:19 AM, Budi wibowo <bwibowo at gmail.com> wrote:

> Thx again, now I move to cdr part. Both cdr in csv and xml are working, but
> I want to store in mysql db. Db structure already created, following the
> wiki doc for mod_cdr. But I don't see the cdr logged into mysql, pressing F8
> also not showing any db query.
> Any help?
> Mod_cdr loaded via modules.conf.xml
> TIA
> Budi wibowo
>

mod_cdr has long since been deprecated. You need to use mod_cdr_csv to write
out the SQL statements and use a cron job to insert them into your database.
Check out this page:
http://wiki.freeswitch.org/wiki/Mod_cdr_csv#Example_SQL_Script_to_create_MySQL_cdr_table_for_default_.22example.22_cdrs

There are some examples of how people are doing this.

-MC
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From anthony.minessale at gmail.com  Mon May 10 16:21:33 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 10 May 2010 18:21:33 -0500
Subject: [Freeswitch-users] blind transfer and outbound socket
In-Reply-To: <4BE88C98.5050107@xpirio.com>
References: <4BE87E88.2000106@xpirio.com>
	<AANLkTikcfzQS6uuSHiCAdllO5ApvAiUT_l1aXMSU3HLT@mail.gmail.com>
	<4BE88515.30105@xpirio.com>
	<AANLkTinftpVgycyYVPLh6-xS9aLPFnReMz3AjAVtIfDh@mail.gmail.com>
	<4BE88C98.5050107@xpirio.com>
Message-ID: <AANLkTim01yFq0PyZMBOy7YnZ73jr9RvGQL6jvQCC4h6b@mail.gmail.com>

I would hardly trust getting an email to tell you if it executed?
did you run your app in the foreground with some debug prints to STDERR to
see if its executing?
did you try using netcat or ngrep to monitor the wire ?


2010/5/10 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>

> yes, and this is my problem
> it looks normal but the scripts are not executed (i send a email in the
> fist line of
> the script and there is no mail send to me here).
> the call is hung up without much comment, the dialplan runs through until
> the hangup -
> without the bridge command executed from the script.
>
> attended transfer is here, with this it works
> http://pastebin.freeswitch.org/12959
>
> the scripts do also work if i call the extension directly.
> the extension 50 also rings if i put the bridge in the dialplan - my only
> problem here
> is that it looks like the script is called but it is never executed.
>
> br
>
>
>
> Anthony Minessale wrote:
>
> > it looks like it works fine to me.
> >
> > the call transfers back to the dialplan where it again executes your
> > socket app and that app either does nothing
> > or hangs up.
> >
> > See line 789 of your pastebin.
> >
> >
> > 2010/5/10 Christian L?schenkohl <christian.loeschenkohl at xpirio.com
> > <mailto:christian.loeschenkohl at xpirio.com>>
> >
> >     the scripts do some database lookups (setting callers name) and do
> >     find the called
> >     user e.g. expand the called 50 to 50 at customer.domain.com
> >     <mailto:50 at customer.domain.com> and finaly do the call bridge.
> >
> >     the transfer itself is done by the phone/softclient with a refer
> >     packet, as far as i can see.
> >
> >     a debug trace is here
> >     http://pastebin.freeswitch.org/12958
> >
> >     br
> >
> >     Anthony Minessale wrote:
> >
> >      > what exactly does your script call to try to make it transfer?
> >      >
> >      > 2010/5/10 Christian L?schenkohl
> >     <christian.loeschenkohl at xpirio.com
> >     <mailto:christian.loeschenkohl at xpirio.com>
> >      > <mailto:christian.loeschenkohl at xpirio.com
> >     <mailto:christian.loeschenkohl at xpirio.com>>>
> >      >
> >      >     hello
> >      >
> >      >     i have a problem with a freeswitch hosted pbx system.
> >      >     when i call in to a extension (e.g. 40) an answer the call
> >     and then
> >      >     do a blind transfer to extension 50 it
> >      >     doesn't work.
> >      >     the outbound socket script do not work or do not get called
> >     in this
> >      >     case (no mail from the script).
> >      >
> >      >     on the other hand if i do a attended transfer it works as it
> >     should
> >      >     (sending mail in the script works).
> >      >     the same context, the same scripts.
> >      >
> >      >     we use snom 320 phones, the sip refer looks normal.
> >      >     blind transfer also doesn't work with a softclient (sjphone)
> >      >
> >      >     where could i start?
> >      >
> >      >     br
> >      >
> >      >     --
> >      >     Ing. Christian L?schenkohl
> >      >     Technische Leitung, Forschung & Entwicklung VoIP
> >      >
> >      >     xpirio
> >      >     Telekommunikation & Service GmbH
> >      >     Lakeside B04
> >      >     9020 Klagenfurt
> >      >     Austria
> >      >
> >      >     T  +43 (0) 5 77 11 - 1000
> >      >     F  +43 (0) 5 77 11 - 1002
> >      >     E  christian.loeschenkohl at xpirio.com
> >     <mailto:christian.loeschenkohl at xpirio.com>
> >      >     <mailto:christian.loeschenkohl at xpirio.com
> >     <mailto:christian.loeschenkohl at xpirio.com>>
> >      >
> >      >     _______________________________________________
> >      >     FreeSWITCH-users mailing list
> >      >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >      >     <mailto:FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
> >      >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >      >
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >      >     http://www.freeswitch.org
> >      >
> >      >
> >      >
> >      >
> >      > --
> >      > 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>
> >     <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> >      > <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >     <mailto:MSN%253Aanthony_minessale at hotmail.com<MSN%25253Aanthony_minessale at hotmail.com>
> >>
> >      > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >     <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> >      > <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >     <mailto:PAYPAL%253Aanthony.minessale at gmail.com<PAYPAL%25253Aanthony.minessale at gmail.com>
> >>
> >      > IRC: irc.freenode.net <http://irc.freenode.net>
> >     <http://irc.freenode.net> #freeswitch
> >      >
> >      > FreeSWITCH Developer Conference
> >      > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >     <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> >      > <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >     <mailto:sip%253A888 at conference.freeswitch.org<sip%25253A888 at conference.freeswitch.org>
> >>
> >      > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >     <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> >      > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >     <mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org<googletalk%25253Aconf%25252B888 at conference.freeswitch.org>
> >>
> >      > pstn:+19193869900
> >      >
> >      >
> >      >
> >
> ------------------------------------------------------------------------
> >      >
> >      > _______________________________________________
> >      > FreeSWITCH-users mailing list
> >      > FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >      > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >      >
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >      > http://www.freeswitch.org
> >
> >
> >     --
> >     Ing. Christian L?schenkohl
> >     Technische Leitung, Forschung & Entwicklung VoIP
> >
> >     xpirio
> >     Telekommunikation & Service GmbH
> >     Lakeside B04
> >     9020 Klagenfurt
> >     Austria
> >
> >     T  +43 (0) 5 77 11 - 1000
> >     F  +43 (0) 5 77 11 - 1002
> >     E  christian.loeschenkohl at xpirio.com
> >     <mailto:christian.loeschenkohl at xpirio.com>
> >
> >     _______________________________________________
> >     FreeSWITCH-users mailing list
> >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     http://www.freeswitch.org
> >
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> > IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> > pstn:+19193869900
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> --
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
>
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
>
> T  +43 (0) 5 77 11 - 1000
> F  +43 (0) 5 77 11 - 1002
> E  christian.loeschenkohl at xpirio.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <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
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From msc at freeswitch.org  Mon May 10 16:39:51 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 10 May 2010 16:39:51 -0700
Subject: [Freeswitch-users] luarun fs_cli output
In-Reply-To: <s2p8fa0f3241005080633ue7f5555axf51c8c02cb15da4@mail.gmail.com>
References: <s2p8fa0f3241005080633ue7f5555axf51c8c02cb15da4@mail.gmail.com>
Message-ID: <AANLkTinE07B3GQSfobI3ZaC4JN5JJq2PV713bgcBq9NK@mail.gmail.com>

On Sat, May 8, 2010 at 6:33 AM, Brian Solon <brian9 at gmail.com> wrote:

> Hi,
>
> This is my first post here so I just wanted to say FreeSWITCH is fantastic.
> I'm using it for an art installation which has been up and running for the
> last week in a gallery in Sligo, Ireland. ( More info here, for the
> curious: http://facebook.com/HelloOperatorIE ).
>
> Just one question for now: how can I see the output of a Lua script when
> launched from fs_cli?
>

freeswitch.consoleLog("INFO","foo bar, dude!\n")
-MC


> -bash-3.2$ pwd
> /opt/freeswitch/scripts
> -bash-3.2$ lua
> Lua 5.1.2  Copyright (C) 1994-2007 Lua.org, PUC-Rio
> > require "test"
> Hello World
> >
>
> I'm probably missing something simple, but when I run it from the CLI all I
> get is "+OK":
>
> freeswitch at internal> luarun test.lua
> +OK
>
> Thanks,
> Brian
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon May 10 16:46:12 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 10 May 2010 16:46:12 -0700
Subject: [Freeswitch-users] Voice to text / speech recognition
In-Reply-To: <09FF0B4B64784D0AA874349E55F2C0C9@dell9400>
References: <alpine.LRH.2.00.1005101227390.20426@osgood.microoffice.com>
	<09FF0B4B64784D0AA874349E55F2C0C9@dell9400>
Message-ID: <AANLkTinEpJxIPc9anUbZEN10UbOxfV0OxSF_ihILi8TC@mail.gmail.com>

You guys need to check out Vestec. Go buy a $25 SDK for your dev server.
Tell them that the FS guys sent you and they'll refund your $25 (check via
snail mail). I haven't had a chance to dig into this one yet but I fully
intend to do so. However, if you guys are working on real solutions to real
world problems then you're in a great position to give Vestec a test drive.
More info: http://www.freeswitch.org/node/252

See, you should visit freeswitch.org more frequently than twice a year! ;)

-MC

On Mon, May 10, 2010 at 3:49 PM, Jan Berger <jan.berger at video24.no> wrote:

> Hi,
>
> I am going to look into Sphinx myself for IVR menus.
>
> Recognition factor is not so much the software as it is the quality of the
> database and how well it is tuned. Tuning you can do yourself with tools
> provided from sphinx + it does exist a separate open source database that I
> believe you can use.
>
> Doing IVR menus is feasible. But, to translate free speech you have a
> challenge called dialects. The fact is that recognition factor is not 100%
> even with people with the same dialect, and many dialects are very far out
> so analyze the language of your clients and make some very accurate
> questions to the vendor to ensure you get what you want.
>
> Jan
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Wellie
> Chao
> Sent: 10. mai 2010 19:09
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Voice to text / speech recognition
>
> I'm looking to provide a feature to transcribe voicemails into text for my
> users. I've looked at SpinVox (which was acquired by Nuance). Their
> licensing method doesn't seem particularly friendly or flexible. They
> charge per user and lock each license to a particular user's actual
> identity. If I have a user named John Smith, I have to buy a license from
> Nuance _for John Smith_. If John Smith stops using the voicemail to text
> feature, but I have another user named Jane Doe who wants to start using
> the voicemail to text feature, I have to buy a new license for Jane Doe.
> Since Nuance makes you pay for a license one year at a time, it gets
> expensive if you have any significant churn. Apart from that, it is just
> annoying to have to register users with another company. I'd like to buy
> 100 or 1000 or 10000 licenses and use them how I see fit and not have to
> bother with registering individual users with a provider of speech
> recognition services.
>
> I am wondering if other FreeSWITCH users have recommendations for good
> voice to text services or software. I'd be happy to consider either a
> service or software. If software, ideally it would be free of course, but
> a reasonable cost would also be acceptable. I know about PocketSphinx, but
> is it really sufficiently high quality as to be useful for transcribing
> voicemails? I don't need 100% accuracy -- probably 80% or 90% would be
> good enough since I will also attach the WAV file and direct my users to
> use the WAV file when in doubt.
>
> Any pointers or tips would be appreciated.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From bwibowo at gmail.com  Mon May 10 17:59:00 2010
From: bwibowo at gmail.com (Budi wibowo)
Date: Tue, 11 May 2010 00:59:00 +0000
Subject: [Freeswitch-users] Mod_cdr
Message-ID: <1188751704-1273539541-cardhu_decombobulator_blackberry.rim.net-1156980273-@bda057.bisx.prodap.on.blackberry>

Hi 
I read from wiki about mod_cdr and it is mention as non functional. My purpose is to log cdr into mysql db, anyboy can share success story about this?


Regards
Budi



From msc at freeswitch.org  Mon May 10 18:08:21 2010
From: msc at freeswitch.org (Michael S Collins)
Date: Mon, 10 May 2010 18:08:21 -0700
Subject: [Freeswitch-users] Mod_cdr
In-Reply-To: <1188751704-1273539541-cardhu_decombobulator_blackberry.rim.net-1156980273-@bda057.bisx.prodap.on.blackberry>
References: <1188751704-1273539541-cardhu_decombobulator_blackberry.rim.net-1156980273-@bda057.bisx.prodap.on.blackberry>
Message-ID: <F6F813AF-2044-4B5F-A246-2C9436FD1153@freeswitch.org>

See my reply earlier today to your previous thread.
-MC

Sent from my iPhone

On May 10, 2010, at 5:59 PM, "Budi wibowo" <bwibowo at gmail.com> wrote:

> Hi
> I read from wiki about mod_cdr and it is mention as non functional.  
> My purpose is to log cdr into mysql db, anyboy can share success  
> story about this?
>
>
> Regards
> Budi
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From bwibowo at gmail.com  Mon May 10 18:13:00 2010
From: bwibowo at gmail.com (Budi wibowo)
Date: Tue, 11 May 2010 01:13:00 +0000
Subject: [Freeswitch-users] outgoing gateway
In-Reply-To: <AANLkTikSvL5MA5omxcbe8rZ_Js6Me0h8LfXyMhQWvQT-@mail.gmail.com>
References: <472BC89E-11F0-49C2-8EA5-23783BC38770@jerris.com><y2z23f91031005071930p3f96278aj9f7db3a5e0df60aa@mail.gmail.com><r2h547ec34a1005080126i9e6f0cche81e1c1118ece6ef@mail.gmail.com><09D47C25-81C0-4815-B477-7595D769EF60@gmail.com><h2s547ec34a1005081716xc09243cq20ec07fda9ba863c@mail.gmail.com><0EC75EB8-42A9-4C56-9DA7-27C8AA642131@gmail.com><v2o547ec34a1005091506m991f371aj9213eb5e4965ea80@mail.gmail.com><o2y547ec34a1005091542p219ff4c6w5b2d64041f511e85@mail.gmail.com><20100510001816.GA4318@jdc.jasonjgw.net><1419115537-1273493954-cardhu_decombobulator_blackberry.rim.net-840701352-@bda057.bisx.prodap.on.blackberry><AANLkTikSvL5MA5omxcbe8rZ_Js6Me0h8LfXyMhQWvQT-@mail.gmail.com>
Message-ID: <1286515355-1273540381-cardhu_decombobulator_blackberry.rim.net-591230050-@bda057.bisx.prodap.on.blackberry>

Thx a lot michael 

-----Original Message-----
From: Michael Collins <msc at freeswitch.org>
Date: Mon, 10 May 2010 15:57:43 
To: <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] outgoing gateway

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




From neilp at cs.stanford.edu  Mon May 10 18:35:42 2010
From: neilp at cs.stanford.edu (Neil Patel)
Date: Mon, 10 May 2010 18:35:42 -0700
Subject: [Freeswitch-users] how do I detect a call has rung once?
In-Reply-To: <AANLkTilVHAdZDgF3a5Mg2HgXrXOg6hkrLKGb9xUAAdik@mail.gmail.com>
References: <u2je5503bdb1005101403u405a5eebo9a2b8c698e9cffaf@mail.gmail.com> 
	<AANLkTinS6No0-1Z1oe9qF6T_9yVedPl3JI-T2OopEoep@mail.gmail.com> 
	<201005101734.09903.sos@sokhapkin.dyndns.org>
	<AANLkTilVHAdZDgF3a5Mg2HgXrXOg6hkrLKGb9xUAAdik@mail.gmail.com>
Message-ID: <AANLkTikqOpUTWkv9izTLM8CKljgexflNPPEVUxiYzw01@mail.gmail.com>

What's the channel variable I monitor for? Is it good practice to poll for
it, or is there another option? Also, does 180 and/or 183 come back for
non-sip calls (I'm dialing out over PRI/openzap).

Thanks,
Neil

On Mon, May 10, 2010 at 2:49 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Unfortunately it's not an assumption.
>
> You have to consider that things are sane.  If they are.  When you place a
> call and you get either a 180 or 183 it denotes that ringing has been
> indicated by the far end.  If you want to get into the gory details they map
> over to a very specific set of ISDN signals that indicate that the phone has
> rung or not both with and without media.  In either case both with and
> without audio if FreeSWITCH gets the early media indication it means someone
> has told us there is ringing.
>
> If things are not sane, you cannot do anything but find another route.
>
>
>
>
>
> On Mon, May 10, 2010 at 4:34 PM, Sergey Okhapkin <sos at sokhapkin.dyndns.org
> > wrote:
>
>> Unfortunately this is a wrong assumption. You can get dead air in early
>> media
>> and SIP error after that.
>>
>> On Monday 10 May 2010, Anthony Minessale wrote:
>> > as soon as you get the first early media signal, that is how you know.
>> >
>> > On Mon, May 10, 2010 at 4:03 PM, Neil Patel <neilp at cs.stanford.edu>
>> wrote:
>> > > I want to initiate a missed call from FS (via lua script). Is there
>> > > anything in the early media I can check to see that the call has rung
>> at
>> > > the endpoint at least once?
>> > >
>> > > Thanks,
>> > > Neil
>> > >
>> > > _______________________________________________
>> > > FreeSWITCH-users mailing list
>> > > FreeSWITCH-users at lists.freeswitch.org
>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > > http://www.freeswitch.org
>> >
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Mon May 10 18:48:57 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 10 May 2010 20:48:57 -0500
Subject: [Freeswitch-users] how do I detect a call has rung once?
In-Reply-To: <AANLkTikqOpUTWkv9izTLM8CKljgexflNPPEVUxiYzw01@mail.gmail.com>
References: <u2je5503bdb1005101403u405a5eebo9a2b8c698e9cffaf@mail.gmail.com>
	<AANLkTinS6No0-1Z1oe9qF6T_9yVedPl3JI-T2OopEoep@mail.gmail.com>
	<201005101734.09903.sos@sokhapkin.dyndns.org>
	<AANLkTilVHAdZDgF3a5Mg2HgXrXOg6hkrLKGb9xUAAdik@mail.gmail.com>
	<AANLkTikqOpUTWkv9izTLM8CKljgexflNPPEVUxiYzw01@mail.gmail.com>
Message-ID: <E08F3AA9-613E-446A-BD22-5B31F38EB78C@freeswitch.org>

yes you will get progressing and alerting but I no not recall which those map to.

/b


Sent from my iPad

On May 10, 2010, at 8:35 PM, Neil Patel <neilp at cs.stanford.edu> wrote:

> What's the channel variable I monitor for? Is it good practice to poll for it, or is there another option? Also, does 180 and/or 183 come back for non-sip calls (I'm dialing out over PRI/openzap).
> 
> Thanks,
> Neil


From dujinfang at gmail.com  Mon May 10 19:19:48 2010
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 11 May 2010 10:19:48 +0800
Subject: [Freeswitch-users] Run FreeSWITCH on iPhone/iPad
Message-ID: <AANLkTinsaq3sx0deoy7sdEuqYoASSJhGXwlUOq_O_LyV@mail.gmail.com>

Hi,


I'd like to know if FS can run on iPhone/iPad, or how hard to make it
happen. I'm interesting to implement a VoIP client on the i platform.

Thanks.

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


From msc at freeswitch.org  Mon May 10 20:10:41 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 10 May 2010 20:10:41 -0700
Subject: [Freeswitch-users] how do I detect a call has rung once?
In-Reply-To: <E08F3AA9-613E-446A-BD22-5B31F38EB78C@freeswitch.org>
References: <u2je5503bdb1005101403u405a5eebo9a2b8c698e9cffaf@mail.gmail.com>
	<AANLkTinS6No0-1Z1oe9qF6T_9yVedPl3JI-T2OopEoep@mail.gmail.com>
	<201005101734.09903.sos@sokhapkin.dyndns.org>
	<AANLkTilVHAdZDgF3a5Mg2HgXrXOg6hkrLKGb9xUAAdik@mail.gmail.com>
	<AANLkTikqOpUTWkv9izTLM8CKljgexflNPPEVUxiYzw01@mail.gmail.com>
	<E08F3AA9-613E-446A-BD22-5B31F38EB78C@freeswitch.org>
Message-ID: <AANLkTimHH6zzA_-Dal1YkHlSL4gkbp7m8TyoFrukBmB3@mail.gmail.com>

Off the top of my head possibly it's ${endpoint_dispostion}
It's been quite a while since I did my openzap stuff...
-MC

On Mon, May 10, 2010 at 6:48 PM, Brian West <brian at freeswitch.org> wrote:

> yes you will get progressing and alerting but I no not recall which those
> map to.
>
> /b
>
>
> Sent from my iPad
>
> On May 10, 2010, at 8:35 PM, Neil Patel <neilp at cs.stanford.edu> wrote:
>
> > What's the channel variable I monitor for? Is it good practice to poll
> for it, or is there another option? Also, does 180 and/or 183 come back for
> non-sip calls (I'm dialing out over PRI/openzap).
> >
> > Thanks,
> > Neil
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Mon May 10 20:11:04 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 10 May 2010 20:11:04 -0700
Subject: [Freeswitch-users] how do I detect a call has rung once?
In-Reply-To: <AANLkTimHH6zzA_-Dal1YkHlSL4gkbp7m8TyoFrukBmB3@mail.gmail.com>
References: <u2je5503bdb1005101403u405a5eebo9a2b8c698e9cffaf@mail.gmail.com>
	<AANLkTinS6No0-1Z1oe9qF6T_9yVedPl3JI-T2OopEoep@mail.gmail.com>
	<201005101734.09903.sos@sokhapkin.dyndns.org>
	<AANLkTilVHAdZDgF3a5Mg2HgXrXOg6hkrLKGb9xUAAdik@mail.gmail.com>
	<AANLkTikqOpUTWkv9izTLM8CKljgexflNPPEVUxiYzw01@mail.gmail.com>
	<E08F3AA9-613E-446A-BD22-5B31F38EB78C@freeswitch.org>
	<AANLkTimHH6zzA_-Dal1YkHlSL4gkbp7m8TyoFrukBmB3@mail.gmail.com>
Message-ID: <AANLkTinHQnCTPTDtxXRl80Pd9aGMs8V2Xws1z5KeMegU@mail.gmail.com>

doh
I mean ${endpoint_disposition}

On Mon, May 10, 2010 at 8:10 PM, Michael Collins <msc at freeswitch.org> wrote:

> Off the top of my head possibly it's ${endpoint_dispostion}
> It's been quite a while since I did my openzap stuff...
> -MC
>
>
> On Mon, May 10, 2010 at 6:48 PM, Brian West <brian at freeswitch.org> wrote:
>
>> yes you will get progressing and alerting but I no not recall which those
>> map to.
>>
>> /b
>>
>>
>> Sent from my iPad
>>
>> On May 10, 2010, at 8:35 PM, Neil Patel <neilp at cs.stanford.edu> wrote:
>>
>> > What's the channel variable I monitor for? Is it good practice to poll
>> for it, or is there another option? Also, does 180 and/or 183 come back for
>> non-sip calls (I'm dialing out over PRI/openzap).
>> >
>> > Thanks,
>> > Neil
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From brian at freeswitch.org  Mon May 10 20:16:50 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 10 May 2010 22:16:50 -0500
Subject: [Freeswitch-users] Run FreeSWITCH on iPhone/iPad
In-Reply-To: <AANLkTinsaq3sx0deoy7sdEuqYoASSJhGXwlUOq_O_LyV@mail.gmail.com>
References: <AANLkTinsaq3sx0deoy7sdEuqYoASSJhGXwlUOq_O_LyV@mail.gmail.com>
Message-ID: <88A46FDA-1C83-43B4-A673-8BDE27F79CCF@freeswitch.org>

I would love to see that.

/b

Sent from my iPad

On May 10, 2010, at 9:19 PM, Seven Du <dujinfang at gmail.com> wrote:

> I'd like to know if FS can run on iPhone/iPad, or how hard to make it
> happen. I'm interesting to implement a VoIP client on the i platform.


From dujinfang at gmail.com  Mon May 10 20:57:53 2010
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 11 May 2010 11:57:53 +0800
Subject: [Freeswitch-users] Run FreeSWITCH on iPhone/iPad
In-Reply-To: <88A46FDA-1C83-43B4-A673-8BDE27F79CCF@freeswitch.org>
References: <AANLkTinsaq3sx0deoy7sdEuqYoASSJhGXwlUOq_O_LyV@mail.gmail.com>
	<88A46FDA-1C83-43B4-A673-8BDE27F79CCF@freeswitch.org>
Message-ID: <AANLkTimyinoIQFdHAncswI-UkIRBv6oN5S2SqLTzkzXW@mail.gmail.com>

I have never developed sth. on the i platform, but rumors said that
program running on iPhone is resource limited(cpu, mem) the the APR
lib used in FS is for cross platform but not aimed for embedded
devices. So I'm collecting informations before start work on it. Also
it maybe need to make portaudio for iPhone work or write a new model.

Anyway, here is an interesting blog page(old though):

http://www.mgamble.ca/blog/2007/10/29/sofia-sip-ported-to-iphone/

2010/5/11 Brian West <brian at freeswitch.org>:
> I would love to see that.
>
> /b
>
> Sent from my iPad
>
> On May 10, 2010, at 9:19 PM, Seven Du <dujinfang at gmail.com> wrote:
>
>> I'd like to know if FS can run on iPhone/iPad, or how hard to make it
>> happen. I'm interesting to implement a VoIP client on the i platform.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



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


From david.ponzone at gmail.com  Mon May 10 22:56:41 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Tue, 11 May 2010 07:56:41 +0200
Subject: [Freeswitch-users] blind transfer and outbound socket
In-Reply-To: <4BE88C98.5050107@xpirio.com>
References: <4BE87E88.2000106@xpirio.com>	<AANLkTikcfzQS6uuSHiCAdllO5ApvAiUT_l1aXMSU3HLT@mail.gmail.com>	<4BE88515.30105@xpirio.com>
	<AANLkTinftpVgycyYVPLh6-xS9aLPFnReMz3AjAVtIfDh@mail.gmail.com>
	<4BE88C98.5050107@xpirio.com>
Message-ID: <3A758DA1-0AD2-401E-9FDF-BD7A4258DC03@gmail.com>

Christian,

if you send the mail using a shell call to "sendmail", it's possible  
it won't work.
Some weeks ago, I and another person have noticed that when you do  
that, sendmail crashes.
I don't know what was the cause and if it is fixed.

Writing to console would be a simpler way to know your script is  
executed.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 11/05/2010 ? 00:45, Christian L?schenkohl a ?crit :

> yes, and this is my problem
> it looks normal but the scripts are not executed (i send a email in  
> the fist line of
> the script and there is no mail send to me here).
> the call is hung up without much comment, the dialplan runs through  
> until the hangup -
> without the bridge command executed from the script.
>
> attended transfer is here, with this it works
> http://pastebin.freeswitch.org/12959
>
> the scripts do also work if i call the extension directly.
> the extension 50 also rings if i put the bridge in the dialplan - my  
> only problem here
> is that it looks like the script is called but it is never executed.
>
> br
>
>
>
> Anthony Minessale wrote:
>
>> it looks like it works fine to me.
>>
>> the call transfers back to the dialplan where it again executes your
>> socket app and that app either does nothing
>> or hangs up.
>>
>> See line 789 of your pastebin.
>>
>>
>> 2010/5/10 Christian L?schenkohl <christian.loeschenkohl at xpirio.com
>> <mailto:christian.loeschenkohl at xpirio.com>>
>>
>>    the scripts do some database lookups (setting callers name) and do
>>    find the called
>>    user e.g. expand the called 50 to 50 at customer.domain.com
>>    <mailto:50 at customer.domain.com> and finaly do the call bridge.
>>
>>    the transfer itself is done by the phone/softclient with a refer
>>    packet, as far as i can see.
>>
>>    a debug trace is here
>>    http://pastebin.freeswitch.org/12958
>>
>>    br
>>
>>    Anthony Minessale wrote:
>>
>>> what exactly does your script call to try to make it transfer?
>>>
>>> 2010/5/10 Christian L?schenkohl
>>    <christian.loeschenkohl at xpirio.com
>>    <mailto:christian.loeschenkohl at xpirio.com>
>>> <mailto:christian.loeschenkohl at xpirio.com
>>    <mailto:christian.loeschenkohl at xpirio.com>>>
>>>
>>>    hello
>>>
>>>    i have a problem with a freeswitch hosted pbx system.
>>>    when i call in to a extension (e.g. 40) an answer the call
>>    and then
>>>    do a blind transfer to extension 50 it
>>>    doesn't work.
>>>    the outbound socket script do not work or do not get called
>>    in this
>>>    case (no mail from the script).
>>>
>>>    on the other hand if i do a attended transfer it works as it
>>    should
>>>    (sending mail in the script works).
>>>    the same context, the same scripts.
>>>
>>>    we use snom 320 phones, the sip refer looks normal.
>>>    blind transfer also doesn't work with a softclient (sjphone)
>>>
>>>    where could i start?
>>>
>>>    br
>>>
>>>    --
>>>    Ing. Christian L?schenkohl
>>>    Technische Leitung, Forschung & Entwicklung VoIP
>>>
>>>    xpirio
>>>    Telekommunikation & Service GmbH
>>>    Lakeside B04
>>>    9020 Klagenfurt
>>>    Austria
>>>
>>>    T  +43 (0) 5 77 11 - 1000
>>>    F  +43 (0) 5 77 11 - 1002
>>>    E  christian.loeschenkohl at xpirio.com
>>    <mailto:christian.loeschenkohl at xpirio.com>
>>>    <mailto:christian.loeschenkohl at xpirio.com
>>    <mailto:christian.loeschenkohl at xpirio.com>>
>>>
>>>    _______________________________________________
>>>    FreeSWITCH-users mailing list
>>>    FreeSWITCH-users at lists.freeswitch.org
>>    <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>    <mailto:FreeSWITCH-users at lists.freeswitch.org
>>    <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>>>    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>    http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
>>    <mailto:MSN%3Aanthony_minessale at hotmail.com>
>>> <mailto:MSN%3Aanthony_minessale at hotmail.com
>>    <mailto:MSN%253Aanthony_minessale at hotmail.com>>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>    <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>>> <mailto:PAYPAL%3Aanthony.minessale at gmail.com
>>    <mailto:PAYPAL%253Aanthony.minessale at gmail.com>>
>>> IRC: irc.freenode.net <http://irc.freenode.net>
>>    <http://irc.freenode.net> #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>    <mailto:sip%3A888 at conference.freeswitch.org>
>>> <mailto:sip%3A888 at conference.freeswitch.org
>>    <mailto:sip%253A888 at conference.freeswitch.org>>
>>> googletalk:conf+888 at conference.freeswitch.org
>>    <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org
>>    <mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org>>
>>> pstn:+19193869900
>>>
>>>
>>>
>>     
>> ------------------------------------------------------------------------
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>    <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>    --
>>    Ing. Christian L?schenkohl
>>    Technische Leitung, Forschung & Entwicklung VoIP
>>
>>    xpirio
>>    Telekommunikation & Service GmbH
>>    Lakeside B04
>>    9020 Klagenfurt
>>    Austria
>>
>>    T  +43 (0) 5 77 11 - 1000
>>    F  +43 (0) 5 77 11 - 1002
>>    E  christian.loeschenkohl at xpirio.com
>>    <mailto:christian.loeschenkohl at xpirio.com>
>>
>>    _______________________________________________
>>    FreeSWITCH-users mailing list
>>    FreeSWITCH-users at lists.freeswitch.org
>>    <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>    http://www.freeswitch.org
>>
>>
>>
>>
>> -- 
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> <mailto:MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> <mailto:sip%3A888 at conference.freeswitch.org>
>> googletalk:conf+888 at conference.freeswitch.org
>> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
>>
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> -- 
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
>
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
>
> T  +43 (0) 5 77 11 - 1000
> F  +43 (0) 5 77 11 - 1002
> E  christian.loeschenkohl at xpirio.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mattdfong at gmail.com  Tue May 11 01:12:55 2010
From: mattdfong at gmail.com (Matthew Fong)
Date: Tue, 11 May 2010 01:12:55 -0700
Subject: [Freeswitch-users] What would cause the UUID of a Channel to Change
	in a Dialplan?
Message-ID: <AANLkTilghkwMw1lOGtKUc1-0wM9i0SCw3BBc7mGqcKi_@mail.gmail.com>

I'm trying to update my application to the latest version of FreeSWITCH. It
uses lua to get a channel's uuid via session:getVariable("uuid") but when I
do this, the uuid mysteriously changes so any uuid_XXX command I execute
fails. At first I thought it might be parking the call (the next dial plan
line), but removing that line did nothing either. I can tell the uuid is
changed by using show channels. Does anyone have any idea what might cause a
channel's uuid to change like this?

Thanks.

--matt
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From jan.berger at video24.no  Tue May 11 01:50:05 2010
From: jan.berger at video24.no (Jan Berger)
Date: Tue, 11 May 2010 10:50:05 +0200
Subject: [Freeswitch-users] Run FreeSWITCH on iPhone/iPad
In-Reply-To: <AANLkTimyinoIQFdHAncswI-UkIRBv6oN5S2SqLTzkzXW@mail.gmail.com>
References: <AANLkTinsaq3sx0deoy7sdEuqYoASSJhGXwlUOq_O_LyV@mail.gmail.com>
	<88A46FDA-1C83-43B4-A673-8BDE27F79CCF@freeswitch.org>
	<AANLkTimyinoIQFdHAncswI-UkIRBv6oN5S2SqLTzkzXW@mail.gmail.com>
Message-ID: <464AEF1A2A2F4620A1D978DC59EFFE68@dell9400>

Your right about limited resources. I can't see why FreeSWITCH should not
run on iPhone/iPad or Symbian or ME, but you would need to limit the modules
down to a bare minimum and start getting greedy on mem-usage. I will however
assume that your main problem is CPU usage - but again if you avoid using
codec's it should work. 

Codec's like G.729/GSM might be a CPU issue. I have not looked into how
iPhone run some of the phone stuff, but my guess is that it's either very
tight assembly or DSP on iPhone doing that???

---

I have considered doing the same using Qt/QtCreator, currently owned by
Nokia because that platform support Windows, Linux, Mac, Solaris, various
Unix/X11, Embedded Linux/ARM, Windows ME and the "blessed" Symbian.

The issue with Symbian is that it's owned by Nokia and used by Sony-Ericsson
on their middle-ware so it's on 60% of the world's mobile devices. Symbian
as such is not a nice OS to develop for. Qt on the other hand is very cool.
 
http://qt.nokia.com

The 2nd reason I am considering QtCreator is because it would give me a
single IDE/framework for a lot of platforms.

Jan

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Seven Du
Sent: 11. mai 2010 05:58
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Run FreeSWITCH on iPhone/iPad

I have never developed sth. on the i platform, but rumors said that
program running on iPhone is resource limited(cpu, mem) the the APR
lib used in FS is for cross platform but not aimed for embedded
devices. So I'm collecting informations before start work on it. Also
it maybe need to make portaudio for iPhone work or write a new model.

Anyway, here is an interesting blog page(old though):

http://www.mgamble.ca/blog/2007/10/29/sofia-sip-ported-to-iphone/

2010/5/11 Brian West <brian at freeswitch.org>:
> I would love to see that.
>
> /b
>
> Sent from my iPad
>
> On May 10, 2010, at 9:19 PM, Seven Du <dujinfang at gmail.com> wrote:
>
>> I'd like to know if FS can run on iPhone/iPad, or how hard to make it
>> happen. I'm interesting to implement a VoIP client on the i platform.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



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

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



From dujinfang at gmail.com  Tue May 11 02:38:43 2010
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 11 May 2010 17:38:43 +0800
Subject: [Freeswitch-users] Run FreeSWITCH on iPhone/iPad
In-Reply-To: <464AEF1A2A2F4620A1D978DC59EFFE68@dell9400>
References: <AANLkTinsaq3sx0deoy7sdEuqYoASSJhGXwlUOq_O_LyV@mail.gmail.com>
	<88A46FDA-1C83-43B4-A673-8BDE27F79CCF@freeswitch.org>
	<AANLkTimyinoIQFdHAncswI-UkIRBv6oN5S2SqLTzkzXW@mail.gmail.com>
	<464AEF1A2A2F4620A1D978DC59EFFE68@dell9400>
Message-ID: <AANLkTilM2s_K8UEJjdC9-2o0SlXX89RwROevXGBPAWRr@mail.gmail.com>

2010/5/11 Jan Berger <jan.berger at video24.no>:
> Your right about limited resources. I can't see why FreeSWITCH should not
> run on iPhone/iPad or Symbian or ME, but you would need to limit the modules
> down to a bare minimum and start getting greedy on mem-usage. I will however
> assume that your main problem is CPU usage - but again if you avoid using
> codec's it should work.
>
> Codec's like G.729/GSM might be a CPU issue. I have not looked into how
> iPhone run some of the phone stuff, but my guess is that it's either very
> tight assembly or DSP on iPhone doing that???
>
> ---
>
> I have considered doing the same using Qt/QtCreator, currently owned by
> Nokia because that platform support Windows, Linux, Mac, Solaris, various
> Unix/X11, Embedded Linux/ARM, Windows ME and the "blessed" Symbian.
>
> The issue with Symbian is that it's owned by Nokia and used by Sony-Ericsson
> on their middle-ware so it's on 60% of the world's mobile devices. Symbian
> as such is not a nice OS to develop for. Qt on the other hand is very cool.
>
> http://qt.nokia.com
>
> The 2nd reason I am considering QtCreator is because it would give me a
> single IDE/framework for a lot of platforms.
>

Sure QT is cool.  FSComm already based on that.

> Jan
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Seven Du
> Sent: 11. mai 2010 05:58
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Run FreeSWITCH on iPhone/iPad
>
> I have never developed sth. on the i platform, but rumors said that
> program running on iPhone is resource limited(cpu, mem) the the APR
> lib used in FS is for cross platform but not aimed for embedded
> devices. So I'm collecting informations before start work on it. Also
> it maybe need to make portaudio for iPhone work or write a new model.
>
> Anyway, here is an interesting blog page(old though):
>
> http://www.mgamble.ca/blog/2007/10/29/sofia-sip-ported-to-iphone/
>
> 2010/5/11 Brian West <brian at freeswitch.org>:
>> I would love to see that.
>>
>> /b
>>
>> Sent from my iPad
>>
>> On May 10, 2010, at 9:19 PM, Seven Du <dujinfang at gmail.com> wrote:
>>
>>> I'd like to know if FS can run on iPhone/iPad, or how hard to make it
>>> happen. I'm interesting to implement a VoIP client on the i platform.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Blog: http://www.dujinfang.com
> Proj: ?http://www.freeswitch.org.cn
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



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


From brian9 at gmail.com  Tue May 11 06:06:29 2010
From: brian9 at gmail.com (Brian Solon)
Date: Tue, 11 May 2010 14:06:29 +0100
Subject: [Freeswitch-users] luarun fs_cli output
In-Reply-To: <AANLkTinE07B3GQSfobI3ZaC4JN5JJq2PV713bgcBq9NK@mail.gmail.com>
References: <s2p8fa0f3241005080633ue7f5555axf51c8c02cb15da4@mail.gmail.com> 
	<AANLkTinE07B3GQSfobI3ZaC4JN5JJq2PV713bgcBq9NK@mail.gmail.com>
Message-ID: <q2w8fa0f3241005110606m179d8b3etf8e2734280d41648@mail.gmail.com>

Thanks!

On 11 May 2010 00:39, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Sat, May 8, 2010 at 6:33 AM, Brian Solon <brian9 at gmail.com> wrote:
>
>> Hi,
>>
>> This is my first post here so I just wanted to say FreeSWITCH is
>> fantastic. I'm using it for an art installation which has been up and
>> running for the last week in a gallery in Sligo, Ireland. ( More info here,
>> for the curious: http://facebook.com/HelloOperatorIE ).
>>
>> Just one question for now: how can I see the output of a Lua script when
>> launched from fs_cli?
>>
>
> freeswitch.consoleLog("INFO","foo bar, dude!\n")
>  -MC
>
>
>> -bash-3.2$ pwd
>> /opt/freeswitch/scripts
>> -bash-3.2$ lua
>> Lua 5.1.2  Copyright (C) 1994-2007 Lua.org, PUC-Rio
>> > require "test"
>> Hello World
>> >
>>
>> I'm probably missing something simple, but when I run it from the CLI all
>> I get is "+OK":
>>
>> freeswitch at internal> luarun test.lua
>> +OK
>>
>> Thanks,
>> Brian
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From rupa at rupa.com  Tue May 11 06:08:37 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 11 May 2010 08:08:37 -0500
Subject: [Freeswitch-users] blind transfer and outbound socket
In-Reply-To: <3A758DA1-0AD2-401E-9FDF-BD7A4258DC03@gmail.com>
References: <4BE87E88.2000106@xpirio.com>
	<AANLkTikcfzQS6uuSHiCAdllO5ApvAiUT_l1aXMSU3HLT@mail.gmail.com>
	<4BE88515.30105@xpirio.com>
	<AANLkTinftpVgycyYVPLh6-xS9aLPFnReMz3AjAVtIfDh@mail.gmail.com>
	<4BE88C98.5050107@xpirio.com>
	<3A758DA1-0AD2-401E-9FDF-BD7A4258DC03@gmail.com>
Message-ID: <AANLkTikpQXPohk12_NENaDJCFrQZy7QET9mTQnLHVNIM@mail.gmail.com>

The cause is a constrained stack space which can't be made larger if you
aren't running FS as super user (you can reduce your stack limit, but once
reduced can't increase it).

I've also run into this in other situations.  I record calls as .wav files
but then convert to mp3.  Initially, I tried to do the conversion via a
system() api in the hangup hook.  But lame would die due to insufficient
stack space.  Instead I now just move the .wav file to a processing
directory where a cron job picks it up, extracts the metadata from the wav
and then converts to mp3 applying the metadata.

You could look at using an alternative sendmail.  Some smtp servers have a
sendmail binary that is very lightweight and would not suffer in a low stack
space environment.  postfix for instance just drops the email into the drop
directory (which is then picked up by the postfix daemon).

The real sendmail on the other hand is going to do everything in-process.
 So it needs to do a lot more work to do that email delivery.

On Tue, May 11, 2010 at 12:56 AM, David Ponzone <david.ponzone at gmail.com>wrote:

> Christian,
>
> if you send the mail using a shell call to "sendmail", it's possible it
> won't work.
> Some weeks ago, I and another person have noticed that when you do that,
> sendmail crashes.
> I don't know what was the cause and if it is fixed.
>
> Writing to console would be a simpler way to know your script is executed.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 11/05/2010 ? 00:45, Christian L?schenkohl a ?crit :
>
> yes, and this is my problem
> it looks normal but the scripts are not executed (i send a email in the
> fist line of
> the script and there is no mail send to me here).
> the call is hung up without much comment, the dialplan runs through until
> the hangup -
> without the bridge command executed from the script.
>
> attended transfer is here, with this it works
> http://pastebin.freeswitch.org/12959
>
> the scripts do also work if i call the extension directly.
> the extension 50 also rings if i put the bridge in the dialplan - my only
> problem here
> is that it looks like the script is called but it is never executed.
>
> br
>
>
>
> Anthony Minessale wrote:
>
> it looks like it works fine to me.
>
>
> the call transfers back to the dialplan where it again executes your
>
> socket app and that app either does nothing
>
> or hangs up.
>
>
> See line 789 of your pastebin.
>
>
>
> 2010/5/10 Christian L?schenkohl <christian.loeschenkohl at xpirio.com
>
> <mailto:christian.loeschenkohl at xpirio.com>>
>
>
>    the scripts do some database lookups (setting callers name) and do
>
>    find the called
>
>    user e.g. expand the called 50 to 50 at customer.domain.com
>
>    <mailto:50 at customer.domain.com> and finaly do the call bridge.
>
>
>    the transfer itself is done by the phone/softclient with a refer
>
>    packet, as far as i can see.
>
>
>    a debug trace is here
>
>    http://pastebin.freeswitch.org/12958
>
>
>    br
>
>
>    Anthony Minessale wrote:
>
>
> what exactly does your script call to try to make it transfer?
>
>
> 2010/5/10 Christian L?schenkohl
>
>    <christian.loeschenkohl at xpirio.com
>
>    <mailto:christian.loeschenkohl at xpirio.com>
>
> <mailto:christian.loeschenkohl at xpirio.com
>
>    <mailto:christian.loeschenkohl at xpirio.com>>>
>
>
>    hello
>
>
>    i have a problem with a freeswitch hosted pbx system.
>
>    when i call in to a extension (e.g. 40) an answer the call
>
>    and then
>
>    do a blind transfer to extension 50 it
>
>    doesn't work.
>
>    the outbound socket script do not work or do not get called
>
>    in this
>
>    case (no mail from the script).
>
>
>    on the other hand if i do a attended transfer it works as it
>
>    should
>
>    (sending mail in the script works).
>
>    the same context, the same scripts.
>
>
>    we use snom 320 phones, the sip refer looks normal.
>
>    blind transfer also doesn't work with a softclient (sjphone)
>
>
>    where could i start?
>
>
>    br
>
>
>    --
>
>    Ing. Christian L?schenkohl
>
>    Technische Leitung, Forschung & Entwicklung VoIP
>
>
>    xpirio
>
>    Telekommunikation & Service GmbH
>
>    Lakeside B04
>
>    9020 Klagenfurt
>
>    Austria
>
>
>    T  +43 (0) 5 77 11 - 1000
>
>    F  +43 (0) 5 77 11 - 1002
>
>    E  christian.loeschenkohl at xpirio.com
>
>    <mailto:christian.loeschenkohl at xpirio.com>
>
>    <mailto:christian.loeschenkohl at xpirio.com
>
>    <mailto:christian.loeschenkohl at xpirio.com>>
>
>
>    _______________________________________________
>
>    FreeSWITCH-users mailing list
>
>    FreeSWITCH-users at lists.freeswitch.org
>
>    <mailto:FreeSWITCH-users at lists.freeswitch.org>
>
>    <mailto:FreeSWITCH-users at lists.freeswitch.org
>
>    <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>
>    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
>
>    UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
>
>    http://www.freeswitch.org
>
>
>
>
>
> --
>
> 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>
>
>    <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
>
> <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
>
>    <mailto:MSN%253Aanthony_minessale at hotmail.com<MSN%25253Aanthony_minessale at hotmail.com>
> >>
>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>
>    <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
>
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
>
>    <mailto:PAYPAL%253Aanthony.minessale at gmail.com<PAYPAL%25253Aanthony.minessale at gmail.com>
> >>
>
> IRC: irc.freenode.net <http://irc.freenode.net>
>
>    <http://irc.freenode.net> #freeswitch
>
>
> FreeSWITCH Developer Conference
>
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>
>    <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
>
> <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
>
>    <mailto:sip%253A888 at conference.freeswitch.org<sip%25253A888 at conference.freeswitch.org>
> >>
>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>
>    <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
>
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
>
>    <mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org<googletalk%25253Aconf%25252B888 at conference.freeswitch.org>
> >>
>
> pstn:+19193869900
>
>
>
>
>    ------------------------------------------------------------------------
>
>
> _______________________________________________
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
>    <mailto:FreeSWITCH-users at lists.freeswitch.org>
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
>
>    UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
>
>    --
>
>    Ing. Christian L?schenkohl
>
>    Technische Leitung, Forschung & Entwicklung VoIP
>
>
>    xpirio
>
>    Telekommunikation & Service GmbH
>
>    Lakeside B04
>
>    9020 Klagenfurt
>
>    Austria
>
>
>    T  +43 (0) 5 77 11 - 1000
>
>    F  +43 (0) 5 77 11 - 1002
>
>    E  christian.loeschenkohl at xpirio.com
>
>    <mailto:christian.loeschenkohl at xpirio.com>
>
>
>    _______________________________________________
>
>    FreeSWITCH-users mailing list
>
>    FreeSWITCH-users at lists.freeswitch.org
>
>    <mailto:FreeSWITCH-users at lists.freeswitch.org>
>
>    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
>    UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
>
>    http://www.freeswitch.org
>
>
>
>
>
> --
>
> Anthony Minessale II
>
>
> FreeSWITCH http://www.freeswitch.org/
>
> ClueCon http://www.cluecon.com/
>
> Twitter: http://twitter.com/FreeSWITCH_wire
>
>
> AIM: anthm
>
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>
> <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
>
> FreeSWITCH Developer Conference
>
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>
> <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
>
> pstn:+19193869900
>
>
>
> ------------------------------------------------------------------------
>
>
> _______________________________________________
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
>
>
> --
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
>
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
>
> T  +43 (0) 5 77 11 - 1000
> F  +43 (0) 5 77 11 - 1002
> E  christian.loeschenkohl at xpirio.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From rupa at rupa.com  Tue May 11 06:10:00 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 11 May 2010 08:10:00 -0500
Subject: [Freeswitch-users] What would cause the UUID of a Channel to
	Change in a Dialplan?
In-Reply-To: <AANLkTilghkwMw1lOGtKUc1-0wM9i0SCw3BBc7mGqcKi_@mail.gmail.com>
References: <AANLkTilghkwMw1lOGtKUc1-0wM9i0SCw3BBc7mGqcKi_@mail.gmail.com>
Message-ID: <AANLkTinhrDNVVLVSlSmvGSSF2kKV8JBsgOl55q6TJL4O@mail.gmail.com>

going through loopback?  Going through fifo?  Maybe giving us your dialplan
would help.  You can also print out the uuid for each step in the dialplan
and narrow down where the uuid changes.

On Tue, May 11, 2010 at 3:12 AM, Matthew Fong <mattdfong at gmail.com> wrote:

> I'm trying to update my application to the latest version of FreeSWITCH. It
> uses lua to get a channel's uuid via session:getVariable("uuid") but when I
> do this, the uuid mysteriously changes so any uuid_XXX command I execute
> fails. At first I thought it might be parking the call (the next dial plan
> line), but removing that line did nothing either. I can tell the uuid is
> changed by using show channels. Does anyone have any idea what might cause a
> channel's uuid to change like this?
>
> Thanks.
>
> --matt
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From brian9 at gmail.com  Tue May 11 06:21:47 2010
From: brian9 at gmail.com (Brian Solon)
Date: Tue, 11 May 2010 14:21:47 +0100
Subject: [Freeswitch-users] luarun fs_cli output
In-Reply-To: <5A3E6BE9-BCA1-4263-8D52-063B2E60D508@jerris.com>
References: <s2p8fa0f3241005080633ue7f5555axf51c8c02cb15da4@mail.gmail.com> 
	<8F5F793A-902D-4932-B17A-7E172E039976@gmail.com>
	<z2o8fa0f3241005091132h7b970f91n8c718fa34bf5cec@mail.gmail.com>
	<5A3E6BE9-BCA1-4263-8D52-063B2E60D508@jerris.com>
Message-ID: <m2h8fa0f3241005110621t44fc5ee6tb1f224beba0c98a@mail.gmail.com>

Hey Mike, I wish I could avail of your offer but it seems airborne
FreeSWITCH ninjas and volcanic ash clouds don't go very well together... :)

On 10 May 2010 19:08, Michael Jerris <mike at jerris.com> wrote:

> I really must be in person at this location in order to answer this
> question properly.  Please send plane tickets and hotel information to my
> email address :D
>
> Mike
>
> On May 9, 2010, at 2:32 PM, Brian Solon wrote:
>
> Thanks, I'll try that.
>
> On 9 May 2010 19:05, David Ponzone <david.ponzone at gmail.com> wrote:
>
>> AFAIK, You can't.
>> You may send the required output to FS console with the required API call
>> (consoleLog).
>>
>
>  Le 08/05/2010 ? 15:33, Brian Solon a ?crit :
>>
>> Hi,
>>
>> This is my first post here so I just wanted to say FreeSWITCH is
>> fantastic. I'm using it for an art installation which has been up and
>> running for the last week in a gallery in Sligo, Ireland. ( More info here,
>> for the curious: http://facebook.com/HelloOperatorIE ).
>>
>> Just one question for now: how can I see the output of a Lua script when
>> launched from fs_cli?
>>
>> -bash-3.2$ pwd
>> /opt/freeswitch/scripts
>> -bash-3.2$ lua
>> Lua 5.1.2  Copyright (C) 1994-2007 Lua.org, PUC-Rio
>> > require "test"
>> Hello World
>> >
>>
>> I'm probably missing something simple, but when I run it from the CLI all
>> I get is "+OK":
>>
>> freeswitch at internal> luarun test.lua
>> +OK
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Tue May 11 08:35:56 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 11 May 2010 10:35:56 -0500
Subject: [Freeswitch-users] how do I detect a call has rung once?
In-Reply-To: <AANLkTinHQnCTPTDtxXRl80Pd9aGMs8V2Xws1z5KeMegU@mail.gmail.com>
References: <u2je5503bdb1005101403u405a5eebo9a2b8c698e9cffaf@mail.gmail.com>
	<AANLkTinS6No0-1Z1oe9qF6T_9yVedPl3JI-T2OopEoep@mail.gmail.com>
	<201005101734.09903.sos@sokhapkin.dyndns.org>
	<AANLkTilVHAdZDgF3a5Mg2HgXrXOg6hkrLKGb9xUAAdik@mail.gmail.com>
	<AANLkTikqOpUTWkv9izTLM8CKljgexflNPPEVUxiYzw01@mail.gmail.com>
	<E08F3AA9-613E-446A-BD22-5B31F38EB78C@freeswitch.org>
	<AANLkTimHH6zzA_-Dal1YkHlSL4gkbp7m8TyoFrukBmB3@mail.gmail.com>
	<AANLkTinHQnCTPTDtxXRl80Pd9aGMs8V2Xws1z5KeMegU@mail.gmail.com>
Message-ID: <AANLkTilcSOp1nfZxSqtndA_HBvKNl1iPrOwJk4uC3P4z@mail.gmail.com>

you don't poll for it you wait for the events.
polling variables is horribly inefficient

On Mon, May 10, 2010 at 10:11 PM, Michael Collins <msc at freeswitch.org>wrote:

> doh
> I mean ${endpoint_disposition}
>
>
> On Mon, May 10, 2010 at 8:10 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Off the top of my head possibly it's ${endpoint_dispostion}
>> It's been quite a while since I did my openzap stuff...
>> -MC
>>
>>
>> On Mon, May 10, 2010 at 6:48 PM, Brian West <brian at freeswitch.org> wrote:
>>
>>> yes you will get progressing and alerting but I no not recall which those
>>> map to.
>>>
>>> /b
>>>
>>>
>>> Sent from my iPad
>>>
>>> On May 10, 2010, at 8:35 PM, Neil Patel <neilp at cs.stanford.edu> wrote:
>>>
>>> > What's the channel variable I monitor for? Is it good practice to poll
>>> for it, or is there another option? Also, does 180 and/or 183 come back for
>>> non-sip calls (I'm dialing out over PRI/openzap).
>>> >
>>> > Thanks,
>>> > Neil
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
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
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From bwibowo at gmail.com  Tue May 11 08:55:46 2010
From: bwibowo at gmail.com (budi wibowo)
Date: Tue, 11 May 2010 22:55:46 +0700
Subject: [Freeswitch-users] about pastebin
Message-ID: <AANLkTil_W6e04K1P25eeUPERzrP5a_jVJ9dnEGCjbZux@mail.gmail.com>

dear all
just wonder about pastebin, anytime i click always ask username and
password. is it the same as freeswitch username and password?
i already create account in fs but cant access pastebin also


TIA

budi
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From jerry.richards at teotech.com  Tue May 11 09:01:07 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 11 May 2010 09:01:07 -0700
Subject: [Freeswitch-users] Efficiency/Performance of mod_xml_curl Versus
	Flat XML Files
Message-ID: <8CF1AD96BF144A8890B50DC1E7EC760A@greyhawk.tonecommander.com>


We are considering converting our flat XML files to database lookups on the
same machine.  It appears mod_xml_curl can be used for this purpose, but we
are wondering how efficient mod_xml_curl is?  We are concerned that it will
increase CPU loading and thus slower response time, especially for dialplan
call routing.  Does anyone have any opinions or recommendations?

Best Regards,
Jerry



From anthony.minessale at gmail.com  Tue May 11 09:08:42 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 11 May 2010 11:08:42 -0500
Subject: [Freeswitch-users] about pastebin
In-Reply-To: <AANLkTil_W6e04K1P25eeUPERzrP5a_jVJ9dnEGCjbZux@mail.gmail.com>
References: <AANLkTil_W6e04K1P25eeUPERzrP5a_jVJ9dnEGCjbZux@mail.gmail.com>
Message-ID: <AANLkTikSjrp1uuQ9SoOwbee4Eh5_uEvcydNiPfEx5QW7@mail.gmail.com>

read the dialog carefully and you will realize the login details

On Tue, May 11, 2010 at 10:55 AM, budi wibowo <bwibowo at gmail.com> wrote:

> dear all
> just wonder about pastebin, anytime i click always ask username and
> password. is it the same as freeswitch username and password?
> i already create account in fs but cant access pastebin also
>
>
> TIA
>
> budi
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <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
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From anthony.minessale at gmail.com  Tue May 11 09:11:31 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 11 May 2010 11:11:31 -0500
Subject: [Freeswitch-users] Efficiency/Performance of mod_xml_curl
	Versus Flat XML Files
In-Reply-To: <8CF1AD96BF144A8890B50DC1E7EC760A@greyhawk.tonecommander.com>
References: <8CF1AD96BF144A8890B50DC1E7EC760A@greyhawk.tonecommander.com>
Message-ID: <AANLkTinUq1YLRZp7ve-Y3ZfYkYIWqnfLPgg5ej1iA34Z@mail.gmail.com>

a nice box, a nice web server and well designed infrastructure

On Tue, May 11, 2010 at 11:01 AM, Jerry Richards <jerry.richards at teotech.com
> wrote:

>
> We are considering converting our flat XML files to database lookups on the
> same machine.  It appears mod_xml_curl can be used for this purpose, but we
> are wondering how efficient mod_xml_curl is?  We are concerned that it will
> increase CPU loading and thus slower response time, especially for dialplan
> call routing.  Does anyone have any opinions or recommendations?
>
> Best Regards,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <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
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From jcasale at activenetwerx.com  Tue May 11 09:19:38 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Tue, 11 May 2010 16:19:38 +0000
Subject: [Freeswitch-users] about pastebin
In-Reply-To: <AANLkTil_W6e04K1P25eeUPERzrP5a_jVJ9dnEGCjbZux@mail.gmail.com>
References: <AANLkTil_W6e04K1P25eeUPERzrP5a_jVJ9dnEGCjbZux@mail.gmail.com>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E9A023A5@prato.activenetwerx.local>

>dear all?
>just wonder about pastebin, anytime i click always ask username and password. is it the same as freeswitch username and password?

Try again, this time, read the login prompt



From brian at freeswitch.org  Tue May 11 09:24:01 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 11 May 2010 11:24:01 -0500
Subject: [Freeswitch-users] about pastebin
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E9A023A5@prato.activenetwerx.local>
References: <AANLkTil_W6e04K1P25eeUPERzrP5a_jVJ9dnEGCjbZux@mail.gmail.com>
	<CA5A491E9DEFBE4CB777DE97E21575E9A023A5@prato.activenetwerx.local>
Message-ID: <D3B512D3-AC5E-4E4B-B0D1-04EAC424AFD4@freeswitch.org>

This shows that you're in a hurry and not aware of your surroundings... Slow down and pay closer attention its only going to bite you late... I'm guilty of being in a hurry at times too.  ;)

/b

On May 11, 2010, at 11:19 AM, Joseph L. Casale wrote:

>> dear all 
>> just wonder about pastebin, anytime i click always ask username and password. is it the same as freeswitch username and password?
> 
> Try again, this time, read the login prompt



From jor3l at foravatars.com  Mon May 10 18:53:55 2010
From: jor3l at foravatars.com (Jor3l Boa)
Date: Mon, 10 May 2010 20:53:55 -0500
Subject: [Freeswitch-users] Socket problem
In-Reply-To: <AANLkTintPWoBG41ZeJtghGVZJSVf72a4OLK15dO3R0Cn@mail.gmail.com>
References: <AANLkTintPWoBG41ZeJtghGVZJSVf72a4OLK15dO3R0Cn@mail.gmail.com>
Message-ID: <AANLkTiltpIJ_eDMzvyaP-wVxZss_HoCCx9rfWtBS41jf@mail.gmail.com>

Hello there, I'm trying to setup FreeSwitch but I'm getting this error
everytime I run it..

*mod_event_socket.c -> Socket Error Could not listen on 127.0.0.1:8021*


Also, an error here:

*mod_xml_curl.c:230 xml_url_fetch() Received HTTP error 0 trying to
> fetch ...*


The question is, is something needed to avoid the 8021 error? I'm running
Ubuntu 10.04 with ufw enabled (port 8021/tcp allowed).

Note: Googled but nothing helps

Thanks in advance
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From lands at freenet.de  Tue May 11 09:36:32 2010
From: lands at freenet.de (lands at freenet.de)
Date: Tue, 11 May 2010 18:36:32 +0200
Subject: [Freeswitch-users] one way audio (sometimes)
Message-ID: <4BE98790.5060605@freenet.de>

Hi,

I have a strange problem with my conference setup on freeswitch.

my setup is as followed:

- freeswitch quick and dirty installation according to
http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install
- freeswitch release: svn-17188
- http://wiki.freeswitch.org/wiki/Examples_JavaScript_Conference_IVR,
- made my DNS SRV records
- account from iptel.org
- "phoner" softphone
- UDP ports are opened inbound from 5060 till 64000 on the firewall,
which is in front of the FS box
- all udp ports outbound are open on the firewall
- the FS box uses a public IP

Well, I can dial the SIP URI. The connection is working. I can hear the
IVR. Well, sometimes at least.
I make a call => I can hear everything.
Seconds later, I make a call with the same client on the same PC =>
audio is coming through.
I cancel the connection and make the call again => there?s no audio
coming through, and so on...
When I do 10 calls within 20 seconds with 4 ones there?s no audio.


According to wireshark the inbound RTP stream is sometimes missing on
some calls.
According to freeswitch debug, the used RTP ports are alway between 5062
and 64000

It sounds like a firewall problem with closed port. But it can?t as the
firewall ports are opened.

Any ideas what else could be wrong?

Chris








From mike at jerris.com  Tue May 11 10:13:21 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 11 May 2010 13:13:21 -0400
Subject: [Freeswitch-users] Socket problem
In-Reply-To: <AANLkTiltpIJ_eDMzvyaP-wVxZss_HoCCx9rfWtBS41jf@mail.gmail.com>
References: <AANLkTintPWoBG41ZeJtghGVZJSVf72a4OLK15dO3R0Cn@mail.gmail.com>
	<AANLkTiltpIJ_eDMzvyaP-wVxZss_HoCCx9rfWtBS41jf@mail.gmail.com>
Message-ID: <4A0DCB25-1013-4303-9BCE-744FB67A8528@jerris.com>

it means something else is already using those ports.  Possibly freeswitch is already running elsewhere on the box or :

netstat -npl | grep 8021

As for the HTTP error 0, I thought this was fixed in git.  Are you using 1.0.6?  try git head and see if you see the same.

Mike

On May 10, 2010, at 9:53 PM, Jor3l Boa wrote:

> Hello there, I'm trying to setup FreeSwitch but I'm getting this error everytime I run it.. 
> 
> mod_event_socket.c -> Socket Error Could not listen on 127.0.0.1:8021
> 
> Also, an error here: 
> 
> mod_xml_curl.c:230 xml_url_fetch() Received HTTP error 0 trying to fetch ...
> 
> The question is, is something needed to avoid the 8021 error? I'm running Ubuntu 10.04 with ufw enabled (port 8021/tcp allowed).
> 
> Note: Googled but nothing helps
> 

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From kris at kriskinc.com  Tue May 11 10:31:18 2010
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Tue, 11 May 2010 13:31:18 -0400
Subject: [Freeswitch-users] OT: Disabling 3pcc in CallManager 7.1
Message-ID: <AANLkTinYGyEHUKwvBvdUDf1wiT57CvQmeBvicR8lUUe7@mail.gmail.com>

Hello everyone,

  I've got a SIP trunk up to a Cisco CallManager 7.1 system.  I've
been able to get it to work by enabling 3pcc in FreeSWITCH but I'd
like to disable 3pcc and be able to have a standard INVITE w/SDP, 183,
200, conversation.  Does anyone have any clue how to configure
CallManager appropriately?

Thanks!

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From ron.freeswitch at mcleodnet.com  Tue May 11 10:35:51 2010
From: ron.freeswitch at mcleodnet.com (Ron McLeod)
Date: Tue, 11 May 2010 10:35:51 -0700
Subject: [Freeswitch-users] short ringback tone when answer
In-Reply-To: <AANLkTikFk61Phm9lKYIh33BfsMFPfgFZrO-rivpJzi3v@mail.gmail.com>
References: <j2g191e3b11005090616ndd0b0ef8g37f60c1f1016ab4f@mail.gmail.com><549CFEF87AEDE841A38E9D15EAB4C04C5576CE4DC7@cooper><y2w191e3b11005091937k6ea8d442i798b95555e11e817@mail.gmail.com>
	<AANLkTikFk61Phm9lKYIh33BfsMFPfgFZrO-rivpJzi3v@mail.gmail.com>
Message-ID: <D0216DB842C84F77999189E6FC9CF38E@fromage>

Tony sent me a Q.931 trace off-list and I thought I would post it here any
case anyone has seen this before.

 

The telco side is including a Signal IE in the SETUP message.  I?m not sure
what this means ? that the telco will play alerting pattern 0 back to the
CPE unless otherwise instructed?

 

Other things in the trace:

  * SETUP is sent again because the dial plan was executing sleep() before
anything triggered a CALL PROCEEDING.

  * The switch PRI stack is reflecting back the Signal IE in the ALERTING,
CONNECT, and RELEASE messages and the telco complains that the IE is not
supported.

 

 

 

RX [08 02 32 11 05 a1 04 03 80 90 a2 18 03 a1 83 97 34 01 40 6c 0a 00 81 XX
XX XX XX 30 36 39 34 70 05 80 36 37 30 30]

Message Type: [05] SETUP

IE: [a1] Sending Complete

IE: [04] Bearer Capability

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

IE: [6c] Calling party number - TON unknown/NPI unknown/Presentation
allowed/User-provided, not screened/XXXX0694

IE: [70] Called party number - TON unknown/NPI unknown/6700

 

 

RX [08 02 32 11 05 a1 04 03 80 90 a2 18 03 a1 83 97 34 01 40 6c 0a 00 81 XX
XX XX XX 30 36 39 34 70 05 80 36 37 30 30]

Message Type: [05] SETUP

IE: [a1] Sending Complete

IE: [04] Bearer Capability

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

IE: [6c] Calling party number - TON unknown/NPI unknown/Presentation
allowed/User-provided, not screened/XXXX0694

IE: [70] Called party number - TON unknown/NPI unknown/6700

 

 

TX [08 02 b2 11 02 04 03 80 90 a2 18 03 a1 83 97]

Message Type: [02] CALL PROCEEDING

IE: [04] Bearer Capability

IE: [18] Channel identification

 

 

TX [08 02 b2 11 01 04 03 80 90 a2 18 03 a1 83 97 34 01 40]

Message Type: [01] ALERTING

IE: [04] Bearer Capability

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

 

 

TX [08 02 b2 11 07 18 03 a1 83 97 34 01 40]

Message Type: [07] CONNECT

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

 

 

RX [08 02 32 11 7d 08 07 82 e3 34 00 00 00 00 14 01 07]

Message Type: [7d] STATUS

IE: [08] Cause -- IE not implemented/Signal

IE: [14] Call State -- call received

 

 

RX [08 02 32 11 7d 08 07 82 e3 34 00 00 00 00 14 01 08]

Message Type: [7d] STATUS

IE: [08] Cause -- IE not implemented/Signal

IE: [14] Call State -- connect request

 

 

RX [08 02 32 11 0f]

Message Type: [0f] CONNECT ACKNOWLEDGE

 

 

RX [08 02 32 11 45 08 02 82 90]

Message Type: [45] DISCONNECT

IE: [08] Cause -- Normal call clearing

 

 

TX [08 02 b2 11 4d 34 01 40]

Message Type: [4d] RELEASE

IE: [34] Signal - Alerting on ? pattern 0

 

 

RX [08 02 32 11 5a 08 07 82 e3 34 00 00 00 00]

Message Type: [5a] RELEASE COMPLETE

IE: [08] Cause -- IE not implemented/Signal

 

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, May 10, 2010 9:15 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

if ringback is not set, it would never play a tone and even if it was, it
would not be played if you explicitly called answer.

Try reversing the answer and the sleep in your dialplan.

 

 

 

On Sun, May 9, 2010 at 9:37 PM, Tony Tin <tony.tin at noahmedia.com.hk> wrote:

Hi Peter,

Thanks for your reply.

The other end is the Telco, I've confirmed with them that they don't play
the tone for us, also I got the Asterisk install in the same box, it doesn't
play the tone. The tone is played exactly when the answer function is
executed, if I put sleep 10 seconds before the answer, the tone will delay
10 seconds too, any idea?

Regards,
Tony





On Mon, May 10, 2010 at 2:33 AM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:

The equipment connected in the other en of the T1 is probably playing these
tones for you, 500ms is probably the time it takes for the called to be
connected successfully.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org
[freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Tony Tin
[tony.tin at noahmedia.com.hk]
Skickat: den 9 maj 2010 15:16
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] short ringback tone when answer


Hi All,

I've just setup a new IVRS with FreeSwitch and a Digium TE220 card (which is
connected to a 4ess T1). Every thing works fine so far except one little
problem that annoys me. Every time I dial in the ivrs, there is a very short
(0.5 second) ringback tone exactly while the "<action
application="answer"/>" line is executed in the dial plan. I want to disable
this ringback tone because the ivrs will answer the call immediately, the
ringback tone is really unnecessary in this case, but I fail to do that. I
try to set the variable "ringback" to change the ringback tone, it's also
not working. Could anyone please help.

I'm using the native pri stack.
The default.xml containts only below lines, and attached is the log file.

      <extension name="testing">
         <condition field="destination_number" expression="^(6.*)$">
               <action application="sleep" data="3000"/>
               <action application="answer"/>
               <action application="sleep" data="3000"/>
         </condition>
      </extension>

Regards,
Tony

!DSPAM:4be6f37432931620317181!

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

 


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




-- 
Anthony Minessale II

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

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

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

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From brian at freeswitch.org  Tue May 11 10:41:13 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 11 May 2010 12:41:13 -0500
Subject: [Freeswitch-users] OT: Disabling 3pcc in CallManager 7.1
In-Reply-To: <AANLkTinYGyEHUKwvBvdUDf1wiT57CvQmeBvicR8lUUe7@mail.gmail.com>
References: <AANLkTinYGyEHUKwvBvdUDf1wiT57CvQmeBvicR8lUUe7@mail.gmail.com>
Message-ID: <4BBED3E8-8D32-4935-83A6-711678F8E828@freeswitch.org>

Nope that thing is EVIL... I haven't the foggiest clue how to do that we have a jira open on this issue.

/b

On May 11, 2010, at 12:31 PM, Kristian Kielhofner wrote:

> Hello everyone,
> 
>  I've got a SIP trunk up to a Cisco CallManager 7.1 system.  I've
> been able to get it to work by enabling 3pcc in FreeSWITCH but I'd
> like to disable 3pcc and be able to have a standard INVITE w/SDP, 183,
> 200, conversation.  Does anyone have any clue how to configure
> CallManager appropriately?
> 
> Thanks!
> 
> -- 



From mike at jerris.com  Tue May 11 11:29:10 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 11 May 2010 14:29:10 -0400
Subject: [Freeswitch-users] short ringback tone when answer
In-Reply-To: <D0216DB842C84F77999189E6FC9CF38E@fromage>
References: <j2g191e3b11005090616ndd0b0ef8g37f60c1f1016ab4f@mail.gmail.com><549CFEF87AEDE841A38E9D15EAB4C04C5576CE4DC7@cooper><y2w191e3b11005091937k6ea8d442i798b95555e11e817@mail.gmail.com>
	<AANLkTikFk61Phm9lKYIh33BfsMFPfgFZrO-rivpJzi3v@mail.gmail.com>
	<D0216DB842C84F77999189E6FC9CF38E@fromage>
Message-ID: <B04A76A2-A198-4E01-95FD-A4BBCF8ED4D5@jerris.com>

which pri stack is this?

Mike

On May 11, 2010, at 1:35 PM, Ron McLeod wrote:

> Tony sent me a Q.931 trace off-list and I thought I would post it here any case anyone has seen this before.
>  
> The telco side is including a Signal IE in the SETUP message.  I?m not sure what this means ? that the telco will play alerting pattern 0 back to the CPE unless otherwise instructed?
>  
> Other things in the trace:
>   * SETUP is sent again because the dial plan was executing sleep() before anything triggered a CALL PROCEEDING.
>   * The switch PRI stack is reflecting back the Signal IE in the ALERTING, CONNECT, and RELEASE messages and the telco complains that the IE is not supported.
>  
>  
>  
> RX [08 02 32 11 05 a1 04 03 80 90 a2 18 03 a1 83 97 34 01 40 6c 0a 00 81 XX XX XX XX 30 36 39 34 70 05 80 36 37 30 30]
> Message Type: [05] SETUP
> IE: [a1] Sending Complete
> IE: [04] Bearer Capability
> IE: [18] Channel identification
> IE: [34] Signal - Alerting on ? pattern 0
> IE: [6c] Calling party number - TON unknown/NPI unknown/Presentation allowed/User-provided, not screened/XXXX0694
> IE: [70] Called party number - TON unknown/NPI unknown/6700
>  
>  
> RX [08 02 32 11 05 a1 04 03 80 90 a2 18 03 a1 83 97 34 01 40 6c 0a 00 81 XX XX XX XX 30 36 39 34 70 05 80 36 37 30 30]
> Message Type: [05] SETUP
> IE: [a1] Sending Complete
> IE: [04] Bearer Capability
> IE: [18] Channel identification
> IE: [34] Signal - Alerting on ? pattern 0
> IE: [6c] Calling party number - TON unknown/NPI unknown/Presentation allowed/User-provided, not screened/XXXX0694
> IE: [70] Called party number - TON unknown/NPI unknown/6700
>  
>  
> TX [08 02 b2 11 02 04 03 80 90 a2 18 03 a1 83 97]
> Message Type: [02] CALL PROCEEDING
> IE: [04] Bearer Capability
> IE: [18] Channel identification
>  
>  
> TX [08 02 b2 11 01 04 03 80 90 a2 18 03 a1 83 97 34 01 40]
> Message Type: [01] ALERTING
> IE: [04] Bearer Capability
> IE: [18] Channel identification
> IE: [34] Signal - Alerting on ? pattern 0
>  
>  
> TX [08 02 b2 11 07 18 03 a1 83 97 34 01 40]
> Message Type: [07] CONNECT
> IE: [18] Channel identification
> IE: [34] Signal - Alerting on ? pattern 0
>  
>  
> RX [08 02 32 11 7d 08 07 82 e3 34 00 00 00 00 14 01 07]
> Message Type: [7d] STATUS
> IE: [08] Cause -- IE not implemented/Signal
> IE: [14] Call State -- call received
>  
>  
> RX [08 02 32 11 7d 08 07 82 e3 34 00 00 00 00 14 01 08]
> Message Type: [7d] STATUS
> IE: [08] Cause -- IE not implemented/Signal
> IE: [14] Call State -- connect request
>  
>  
> RX [08 02 32 11 0f]
> Message Type: [0f] CONNECT ACKNOWLEDGE
>  
>  
> RX [08 02 32 11 45 08 02 82 90]
> Message Type: [45] DISCONNECT
> IE: [08] Cause -- Normal call clearing
>  
>  
> TX [08 02 b2 11 4d 34 01 40]
> Message Type: [4d] RELEASE
> IE: [34] Signal - Alerting on ? pattern 0
>  
>  
> RX [08 02 32 11 5a 08 07 82 e3 34 00 00 00 00]
> Message Type: [5a] RELEASE COMPLETE
> IE: [08] Cause -- IE not implemented/Signal
>  
>  
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale
> Sent: Monday, May 10, 2010 9:15 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] short ringback tone when answer
>  
> if ringback is not set, it would never play a tone and even if it was, it would not be played if you explicitly called answer.
> Try reversing the answer and the sleep in your dialplan.
>  
>  
>  
> 
> On Sun, May 9, 2010 at 9:37 PM, Tony Tin <tony.tin at noahmedia.com.hk> wrote:
> Hi Peter,
> 
> Thanks for your reply.
> 
> The other end is the Telco, I've confirmed with them that they don't play the tone for us, also I got the Asterisk install in the same box, it doesn't play the tone. The tone is played exactly when the answer function is executed, if I put sleep 10 seconds before the answer, the tone will delay 10 seconds too, any idea?
> 
> Regards,
> Tony
> 
> 
> 
> On Mon, May 10, 2010 at 2:33 AM, Peter Olsson <peter.olsson at visionutveckling.se> wrote:
> The equipment connected in the other en of the T1 is probably playing these tones for you, 500ms is probably the time it takes for the called to be connected successfully.
> 
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Tony Tin [tony.tin at noahmedia.com.hk]
> Skickat: den 9 maj 2010 15:16
> Till: freeswitch-users at lists.freeswitch.org
> ?mne: [Freeswitch-users] short ringback tone when answer
> 
> Hi All,
> 
> I've just setup a new IVRS with FreeSwitch and a Digium TE220 card (which is connected to a 4ess T1). Every thing works fine so far except one little problem that annoys me. Every time I dial in the ivrs, there is a very short (0.5 second) ringback tone exactly while the "<action application="answer"/>" line is executed in the dial plan. I want to disable this ringback tone because the ivrs will answer the call immediately, the ringback tone is really unnecessary in this case, but I fail to do that. I try to set the variable "ringback" to change the ringback tone, it's also not working. Could anyone please help.
> 
> I'm using the native pri stack.
> The default.xml containts only below lines, and attached is the log file.
> 
>       <extension name="testing">
>          <condition field="destination_number" expression="^(6.*)$">
>                <action application="sleep" data="3000"/>
>                <action application="answer"/>
>                <action application="sleep" data="3000"/>
>          </condition>
>       </extension>
> 
> Regards,
> Tony
> 
> !DSPAM:4be6f37432931620317181!
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>  
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From kris at kriskinc.com  Tue May 11 11:29:54 2010
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Tue, 11 May 2010 14:29:54 -0400
Subject: [Freeswitch-users] OT: Disabling 3pcc in CallManager 7.1
In-Reply-To: <4BBED3E8-8D32-4935-83A6-711678F8E828@freeswitch.org>
References: <AANLkTinYGyEHUKwvBvdUDf1wiT57CvQmeBvicR8lUUe7@mail.gmail.com>
	<4BBED3E8-8D32-4935-83A6-711678F8E828@freeswitch.org>
Message-ID: <AANLkTikMaTYX-6_BRBFjon6Q-f3-DaMYyjr3foLoZ8KC@mail.gmail.com>

Someone from Voiceops pointed me in the right direction.  I'm waiting
to confirm but this might be the ticket:

"Media Termination Point Required" is checked "on" in the SIP trunk
configuration window.

That should force an SDP in the initial INVITE...

On Tue, May 11, 2010 at 1:41 PM, Brian West <brian at freeswitch.org> wrote:
> Nope that thing is EVIL... I haven't the foggiest clue how to do that we have a jira open on this issue.
>
> /b
>

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From brian at freeswitch.org  Tue May 11 11:37:26 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 11 May 2010 13:37:26 -0500
Subject: [Freeswitch-users] OT: Disabling 3pcc in CallManager 7.1
In-Reply-To: <AANLkTikMaTYX-6_BRBFjon6Q-f3-DaMYyjr3foLoZ8KC@mail.gmail.com>
References: <AANLkTinYGyEHUKwvBvdUDf1wiT57CvQmeBvicR8lUUe7@mail.gmail.com>
	<4BBED3E8-8D32-4935-83A6-711678F8E828@freeswitch.org>
	<AANLkTikMaTYX-6_BRBFjon6Q-f3-DaMYyjr3foLoZ8KC@mail.gmail.com>
Message-ID: <5EB49AC9-3C7D-403A-84CF-E4F6A89A2C51@freeswitch.org>

We need to wiki that if at all possible once you confirm that it works.

/b

On May 11, 2010, at 1:29 PM, Kristian Kielhofner wrote:

> Someone from Voiceops pointed me in the right direction.  I'm waiting
> to confirm but this might be the ticket:
> 
> "Media Termination Point Required" is checked "on" in the SIP trunk
> configuration window.
> 
> That should force an SDP in the initial INVITE...



From msc at freeswitch.org  Tue May 11 12:01:00 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 11 May 2010 12:01:00 -0700
Subject: [Freeswitch-users] OT: Disabling 3pcc in CallManager 7.1
In-Reply-To: <AANLkTikMaTYX-6_BRBFjon6Q-f3-DaMYyjr3foLoZ8KC@mail.gmail.com>
References: <AANLkTinYGyEHUKwvBvdUDf1wiT57CvQmeBvicR8lUUe7@mail.gmail.com>
	<4BBED3E8-8D32-4935-83A6-711678F8E828@freeswitch.org>
	<AANLkTikMaTYX-6_BRBFjon6Q-f3-DaMYyjr3foLoZ8KC@mail.gmail.com>
Message-ID: <AANLkTinrdi-TfLxWV8jfI4Bc-z1DdI_ebx-vqV59HTxO@mail.gmail.com>

KK,
Thanks, as usual, for sharing the knowledge that you so diligently earn
through hard work. Please keep up the good work. :)

Also, we are booking speakers for ClueCon... we'd love to have you back to
talk about something interesting. Let us know!
Thanks,
MC

On Tue, May 11, 2010 at 11:29 AM, Kristian Kielhofner <kris at kriskinc.com>wrote:

> Someone from Voiceops pointed me in the right direction.  I'm waiting
> to confirm but this might be the ticket:
>
> "Media Termination Point Required" is checked "on" in the SIP trunk
> configuration window.
>
> That should force an SDP in the initial INVITE...
>
> On Tue, May 11, 2010 at 1:41 PM, Brian West <brian at freeswitch.org> wrote:
> > Nope that thing is EVIL... I haven't the foggiest clue how to do that we
> have a jira open on this issue.
> >
> > /b
> >
>
> --
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jor3l at foravatars.com  Tue May 11 13:05:13 2010
From: jor3l at foravatars.com (Jor3l Boa)
Date: Tue, 11 May 2010 15:05:13 -0500
Subject: [Freeswitch-users] Socket problem
In-Reply-To: <4A0DCB25-1013-4303-9BCE-744FB67A8528@jerris.com>
References: <AANLkTintPWoBG41ZeJtghGVZJSVf72a4OLK15dO3R0Cn@mail.gmail.com>
	<AANLkTiltpIJ_eDMzvyaP-wVxZss_HoCCx9rfWtBS41jf@mail.gmail.com>
	<4A0DCB25-1013-4303-9BCE-744FB67A8528@jerris.com>
Message-ID: <AANLkTil0EoBIXqJVs3oc2v5q1-VCBfUNRpV7sWhZWK_t@mail.gmail.com>

Thanks, got it working with the last git :)

2010/5/11 Michael Jerris <mike at jerris.com>

> it means something else is already using those ports.  Possibly freeswitch
> is already running elsewhere on the box or :
>
> netstat -npl | grep 8021
>
> As for the HTTP error 0, I thought this was fixed in git.  Are you using
> 1.0.6?  try git head and see if you see the same.
>
> Mike
>
> On May 10, 2010, at 9:53 PM, Jor3l Boa wrote:
>
> Hello there, I'm trying to setup FreeSwitch but I'm getting this error
> everytime I run it..
>
> *mod_event_socket.c -> Socket Error Could not listen on 127.0.0.1:8021*
>
>
> Also, an error here:
>
> *mod_xml_curl.c:230 xml_url_fetch() Received HTTP error 0 trying to
>> fetch ...*
>
>
> The question is, is something needed to avoid the 8021 error? I'm running
> Ubuntu 10.04 with ufw enabled (port 8021/tcp allowed).
>
> Note: Googled but nothing helps
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From vmknott at gmail.com  Tue May 11 13:07:34 2010
From: vmknott at gmail.com (VM Knott)
Date: Tue, 11 May 2010 16:07:34 -0400
Subject: [Freeswitch-users] In-band vs. Out-of-band DTMF
Message-ID: <AANLkTing5v0xPtghphOVKjAPEZdU5dAiCaEePzIIWq_V@mail.gmail.com>

I'm trying to test DTMF signalling using in-band DTMF.
Currently my application monitors the signals via the ESL socket,
which works great for out-of-band DTMF.

My problem is that when I run my current test using in-band, I'm using
a phone line that I know supports out-of band DTMF, so I'm not sure
that the data I'm seeing in the ESL socket is out-of-band or in-band.

Does anyone know:

a) if in-band signalling appears the same across the ESL socket?

b) if I should be looking somewhere specific for the in-band signals.

c) is there a way to disable out-of-band signalling so I can test only
the in-band?

Any assistance would be greatly appreciated.

Victor


From mark.maly at molcs.org  Tue May 11 13:12:03 2010
From: mark.maly at molcs.org (Mark Maly)
Date: Tue, 11 May 2010 15:12:03 -0500
Subject: [Freeswitch-users] Aastra and SCA
In-Reply-To: <28B636AB-D221-44C5-BDA4-1D12AC44748B@freeswitch.org>
References: <021d01cae8d2$fc999420$f5ccbc60$@maly@molcs.org>	<2A6C7EC2-913D-417E-ADD6-E3A72C6AA6E6@freeswitch.org>	<003b01caeb24$895c7b90$9c1572b0$@maly@molcs.org>
	<28B636AB-D221-44C5-BDA4-1D12AC44748B@freeswitch.org>
Message-ID: <00a701caf146$38f02f30$aad08d90$@maly@molcs.org>

FYI-

 

Got a ticket w/Aastra and they said, 

 

"We have just released a new GA firmware that I would like for you to load
onto one of your phones to see if it resolves the issue. You can download
this from our website at www.aastratelecom.com/support and click on Download
Area and select the 6731i. The firmware version you are looking for is 2.6
and it is listed under Current Software Release.

 

Please load that on a test phone and let us know the results."

 

I'll let you know,

Mark

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Monday, May 03, 2010 8:00 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Aastra and SCA

 

Nope but we have exchanged some emails about it back and forth and some beta
firmware where they did half way fix it... but seems you need the call-info
header on ALL packets associated with the dialog of that call.  They are
still missing a few.

 

/b

 

On May 3, 2010, at 7:55 PM, Mark Maly wrote:





Brian,

 

After my first reply I did receive a msg from Aastra support and whether FS
had reported the problem.   The contact wondered whether they [FS] had been
given any "reference number" related to the problem. 

 

Would love to help and would pass ref num if you had one.

 

Mark

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Sunday, May 02, 2010 11:14 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Aastra and SCA

 

Give up their exists a bug that prevents it from working on any aastra.
Polycom and Cisco SPA work flawless.

 

/b

Sent from my iPad


On Apr 30, 2010, at 9:07 PM, "Mark Maly" <mark.maly at molcs.org> wrote:

Hi,

 

I've tried to patiently figure this out by reading the wiki and this list.
Unfortunately, I've been unable to get it right. 

 

I have 2 Aastra 6731is and a 51i and trying to get SCA working.  I'm
experiencing problems similar to the Cisco thread from last month - outgoing
calls implement SCA well.  Incoming calls ring all lines and appearances
work, but when one phone is answered, the line appearances are removed from
the remaining phones.  I am not attempting to use any DNS.

 

My configuration has all three phones plus FS on a local LAN.  Nothing too
fancy.  Each line is configured for Broadsoft SCA and SCA bridging  is
enabled globally for the phones.

 

Trying to update/replace an old phone at my church.

 

Any help would be greatly appreciated.

 

Thanks, Mark

Mark.maly at molcs.org

 

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

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

 

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From kris at kriskinc.com  Tue May 11 13:22:24 2010
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Tue, 11 May 2010 16:22:24 -0400
Subject: [Freeswitch-users] OT: Disabling 3pcc in CallManager 7.1
In-Reply-To: <AANLkTinrdi-TfLxWV8jfI4Bc-z1DdI_ebx-vqV59HTxO@mail.gmail.com>
References: <AANLkTinYGyEHUKwvBvdUDf1wiT57CvQmeBvicR8lUUe7@mail.gmail.com>
	<4BBED3E8-8D32-4935-83A6-711678F8E828@freeswitch.org>
	<AANLkTikMaTYX-6_BRBFjon6Q-f3-DaMYyjr3foLoZ8KC@mail.gmail.com>
	<AANLkTinrdi-TfLxWV8jfI4Bc-z1DdI_ebx-vqV59HTxO@mail.gmail.com>
Message-ID: <s2l4d15ff861005111322m675b61daxac9631a9af573b8d@mail.gmail.com>

Haha thanks MC but the real credit goes to Mark Holloway on Voiceops
for pointing me in the right direction.

I have tested this with CUCM 7.1 and confirmed it works.  Where should
this be documented?

I'll see if I can come up with anything "interesting" for ClueCon...

On Tue, May 11, 2010 at 3:01 PM, Michael Collins <msc at freeswitch.org> wrote:
> KK,
> Thanks, as usual, for sharing the knowledge that you so diligently earn
> through hard work. Please keep up the good work. :)
>
> Also, we are booking speakers for ClueCon... we'd love to have you back to
> talk about something interesting. Let us know!
> Thanks,
> MC
>

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From msc at freeswitch.org  Tue May 11 14:12:46 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 11 May 2010 14:12:46 -0700
Subject: [Freeswitch-users] OT: Disabling 3pcc in CallManager 7.1
In-Reply-To: <s2l4d15ff861005111322m675b61daxac9631a9af573b8d@mail.gmail.com>
References: <AANLkTinYGyEHUKwvBvdUDf1wiT57CvQmeBvicR8lUUe7@mail.gmail.com>
	<4BBED3E8-8D32-4935-83A6-711678F8E828@freeswitch.org>
	<AANLkTikMaTYX-6_BRBFjon6Q-f3-DaMYyjr3foLoZ8KC@mail.gmail.com>
	<AANLkTinrdi-TfLxWV8jfI4Bc-z1DdI_ebx-vqV59HTxO@mail.gmail.com>
	<s2l4d15ff861005111322m675b61daxac9631a9af573b8d@mail.gmail.com>
Message-ID: <AANLkTik8jXTWta7Wq9NgJQ-rTac7vxwjxjvdInNsAXvw@mail.gmail.com>

On Tue, May 11, 2010 at 1:22 PM, Kristian Kielhofner <kris at kriskinc.com>wrote:

> Haha thanks MC but the real credit goes to Mark Holloway on Voiceops
> for pointing me in the right direction.
>
> I have tested this with CUCM 7.1 and confirmed it works.  Where should
> this be documented?
>
Since you're so gracious:
http://wiki.freeswitch.org/wiki/Cisco_Call_Manager


>
> I'll see if I can come up with anything "interesting" for ClueCon...
>
> Sounds good. If you've done anything new with Recqual that would be fine,
or some other interesting VoIP app...
-MC
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From kenfulmer at icstechnologysolutions.com  Tue May 11 14:28:18 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Tue, 11 May 2010 16:28:18 -0500
Subject: [Freeswitch-users] OT: Disabling 3pcc in CallManager 7.1
In-Reply-To: <AANLkTikMaTYX-6_BRBFjon6Q-f3-DaMYyjr3foLoZ8KC@mail.gmail.com>
References: <AANLkTinYGyEHUKwvBvdUDf1wiT57CvQmeBvicR8lUUe7@mail.gmail.com>	<4BBED3E8-8D32-4935-83A6-711678F8E828@freeswitch.org>
	<AANLkTikMaTYX-6_BRBFjon6Q-f3-DaMYyjr3foLoZ8KC@mail.gmail.com>
Message-ID: <015901caf150$e02f62c0$a08e2840$@com>

MTP required on the Cisco SIP trunk will turn on "early media negotiation".
This means it will send an SDP with its initial Invite, rather than waiting
on the other side to send SDP. 

Hope this helps.

Ken

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kristian
Kielhofner
Sent: Tuesday, May 11, 2010 1:30 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] OT: Disabling 3pcc in CallManager 7.1

Someone from Voiceops pointed me in the right direction.  I'm waiting
to confirm but this might be the ticket:

"Media Termination Point Required" is checked "on" in the SIP trunk
configuration window.

That should force an SDP in the initial INVITE...

On Tue, May 11, 2010 at 1:41 PM, Brian West <brian at freeswitch.org> wrote:
> Nope that thing is EVIL... I haven't the foggiest clue how to do that we
have a jira open on this issue.
>
> /b
>

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com

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



From ron.freeswitch at mcleodnet.com  Tue May 11 14:30:07 2010
From: ron.freeswitch at mcleodnet.com (Ron McLeod)
Date: Tue, 11 May 2010 14:30:07 -0700
Subject: [Freeswitch-users] short ringback tone when answer
In-Reply-To: <B04A76A2-A198-4E01-95FD-A4BBCF8ED4D5@jerris.com>
References: <j2g191e3b11005090616ndd0b0ef8g37f60c1f1016ab4f@mail.gmail.com><549CFEF87AEDE841A38E9D15EAB4C04C5576CE4DC7@cooper><y2w191e3b11005091937k6ea8d442i798b95555e11e817@mail.gmail.com><AANLkTikFk61Phm9lKYIh33BfsMFPfgFZrO-rivpJzi3v@mail.gmail.com><D0216DB842C84F77999189E6FC9CF38E@fromage>
	<B04A76A2-A198-4E01-95FD-A4BBCF8ED4D5@jerris.com>
Message-ID: <6265B4F0F9F949759F3154920D7F4416@fromage>

FreeSWITCH ISDN.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Jerris
Sent: Tuesday, May 11, 2010 11:29 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

which pri stack is this?

 

Mike

 

On May 11, 2010, at 1:35 PM, Ron McLeod wrote:





Tony sent me a Q.931 trace off-list and I thought I would post it here any
case anyone has seen this before.

 

The telco side is including a Signal IE in the SETUP message.  I?m not sure
what this means ? that the telco will play alerting pattern 0 back to the
CPE unless otherwise instructed?

 

Other things in the trace:

  * SETUP is sent again because the dial plan was executing sleep() before
anything triggered a CALL PROCEEDING.

  * The switch PRI stack is reflecting back the Signal IE in the ALERTING,
CONNECT, and RELEASE messages and the telco complains that the IE is not
supported.

 

 

 

RX [08 02 32 11 05 a1 04 03 80 90 a2 18 03 a1 83 97 34 01 40 6c 0a 00 81 XX
XX XX XX 30 36 39 34 70 05 80 36 37 30 30]

Message Type: [05] SETUP

IE: [a1] Sending Complete

IE: [04] Bearer Capability

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

IE: [6c] Calling party number - TON unknown/NPI unknown/Presentation
allowed/User-provided, not screened/XXXX0694

IE: [70] Called party number - TON unknown/NPI unknown/6700

 

 

RX [08 02 32 11 05 a1 04 03 80 90 a2 18 03 a1 83 97 34 01 40 6c 0a 00 81 XX
XX XX XX 30 36 39 34 70 05 80 36 37 30 30]

Message Type: [05] SETUP

IE: [a1] Sending Complete

IE: [04] Bearer Capability

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

IE: [6c] Calling party number - TON unknown/NPI unknown/Presentation
allowed/User-provided, not screened/XXXX0694

IE: [70] Called party number - TON unknown/NPI unknown/6700

 

 

TX [08 02 b2 11 02 04 03 80 90 a2 18 03 a1 83 97]

Message Type: [02] CALL PROCEEDING

IE: [04] Bearer Capability

IE: [18] Channel identification

 

 

TX [08 02 b2 11 01 04 03 80 90 a2 18 03 a1 83 97 34 01 40]

Message Type: [01] ALERTING

IE: [04] Bearer Capability

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

 

 

TX [08 02 b2 11 07 18 03 a1 83 97 34 01 40]

Message Type: [07] CONNECT

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

 

 

RX [08 02 32 11 7d 08 07 82 e3 34 00 00 00 00 14 01 07]

Message Type: [7d] STATUS

IE: [08] Cause -- IE not implemented/Signal

IE: [14] Call State -- call received

 

 

RX [08 02 32 11 7d 08 07 82 e3 34 00 00 00 00 14 01 08]

Message Type: [7d] STATUS

IE: [08] Cause -- IE not implemented/Signal

IE: [14] Call State -- connect request

 

 

RX [08 02 32 11 0f]

Message Type: [0f] CONNECT ACKNOWLEDGE

 

 

RX [08 02 32 11 45 08 02 82 90]

Message Type: [45] DISCONNECT

IE: [08] Cause -- Normal call clearing

 

 

TX [08 02 b2 11 4d 34 01 40]

Message Type: [4d] RELEASE

IE: [34] Signal - Alerting on ? pattern 0

 

 

RX [08 02 32 11 5a 08 07 82 e3 34 00 00 00 00]

Message Type: [5a] RELEASE COMPLETE

IE: [08] Cause -- IE not implemented/Signal

 

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, May 10, 2010 9:15 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

if ringback is not set, it would never play a tone and even if it was, it
would not be played if you explicitly called answer.

Try reversing the answer and the sleep in your dialplan.

 

 

 

On Sun, May 9, 2010 at 9:37 PM, Tony Tin <tony.tin at noahmedia.com.hk> wrote:

Hi Peter,

Thanks for your reply.

The other end is the Telco, I've confirmed with them that they don't play
the tone for us, also I got the Asterisk install in the same box, it doesn't
play the tone. The tone is played exactly when the answer function is
executed, if I put sleep 10 seconds before the answer, the tone will delay
10 seconds too, any idea?

Regards,
Tony






On Mon, May 10, 2010 at 2:33 AM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:

The equipment connected in the other en of the T1 is probably playing these
tones for you, 500ms is probably the time it takes for the called to be
connected successfully.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org
[freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Tony Tin
[tony.tin at noahmedia.com.hk]
Skickat: den 9 maj 2010 15:16
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] short ringback tone when answer


Hi All,

I've just setup a new IVRS with FreeSwitch and a Digium TE220 card (which is
connected to a 4ess T1). Every thing works fine so far except one little
problem that annoys me. Every time I dial in the ivrs, there is a very short
(0.5 second) ringback tone exactly while the "<action
application="answer"/>" line is executed in the dial plan. I want to disable
this ringback tone because the ivrs will answer the call immediately, the
ringback tone is really unnecessary in this case, but I fail to do that. I
try to set the variable "ringback" to change the ringback tone, it's also
not working. Could anyone please help.

I'm using the native pri stack.
The default.xml containts only below lines, and attached is the log file.

      <extension name="testing">
         <condition field="destination_number" expression="^(6.*)$">
               <action application="sleep" data="3000"/>
               <action application="answer"/>
               <action application="sleep" data="3000"/>
         </condition>
      </extension>

Regards,
Tony

!DSPAM:4be6f37432931620317181!

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

 


_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org <http://www.freeswitch.org/> 




-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
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<mailto:sip%3A888 at conference.freeswitch.org> 
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From fsmail at conspiracy.net  Tue May 11 14:39:28 2010
From: fsmail at conspiracy.net (paul bilke)
Date: Tue, 11 May 2010 16:39:28 -0500
Subject: [Freeswitch-users] Freeswitch Scripting strangeness
Message-ID: <4BE9CE90.7050805@conspiracy.net>

I am working on a system that invokes a LUA script of the dial plan to handle interaction on the call.
All is fine when running a single instance of the script (one call).  The issue is when there is more than one call things become erratic.
Right now the script opens a log file who's name is based on the date of the call and its DNIS using io.open with append.
The script then writes lines to file as it runs detailing its actions.  The same log function also uses posix.syslog to log the same information
via syslog.  All is fine when there is one call up.
If there is more than one concurrent on two different DNIS's both log files a present but have output that was created by the other call.
The syslog is sane but not the file of logging information.  The time stamp will jump backward and pickup with information from the other call
(can tell via the UUID)
It is my assumption that each call should spin up a thread and they should share no data/variables/handles with other calls in the system.  Looking at
the logs indicates there is some issue but logging is not the only issue since it seems that call events updating state information is affecting the other
call also.
If my assumption on a isolated thread per call is incorrect let me know.
I inherited this code and suspect that running an application out of the dialplan script is unconventional.
Paul





From mike at jerris.com  Tue May 11 15:52:45 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 11 May 2010 18:52:45 -0400
Subject: [Freeswitch-users] short ringback tone when answer
In-Reply-To: <6265B4F0F9F949759F3154920D7F4416@fromage>
References: <j2g191e3b11005090616ndd0b0ef8g37f60c1f1016ab4f@mail.gmail.com><549CFEF87AEDE841A38E9D15EAB4C04C5576CE4DC7@cooper><y2w191e3b11005091937k6ea8d442i798b95555e11e817@mail.gmail.com><AANLkTikFk61Phm9lKYIh33BfsMFPfgFZrO-rivpJzi3v@mail.gmail.com><D0216DB842C84F77999189E6FC9CF38E@fromage>
	<B04A76A2-A198-4E01-95FD-A4BBCF8ED4D5@jerris.com>
	<6265B4F0F9F949759F3154920D7F4416@fromage>
Message-ID: <EFB02571-272A-4F3D-AD91-4B78AAA2B5E9@jerris.com>

Ron-

	Catch me online and I can work on a patch to address this for you if I can get remote access to test.

Mike

On May 11, 2010, at 5:30 PM, Ron McLeod wrote:

> FreeSWITCH ISDN.
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
> Sent: Tuesday, May 11, 2010 11:29 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] short ringback tone when answer
>  
> which pri stack is this?
>  
> Mike
>  
> On May 11, 2010, at 1:35 PM, Ron McLeod wrote:
> 
> 
> Tony sent me a Q.931 trace off-list and I thought I would post it here any case anyone has seen this before.
>  
> The telco side is including a Signal IE in the SETUP message.  I?m not sure what this means ? that the telco will play alerting pattern 0 back to the CPE unless otherwise instructed?
>  
> Other things in the trace:
>   * SETUP is sent again because the dial plan was executing sleep() before anything triggered a CALL PROCEEDING.
>   * The switch PRI stack is reflecting back the Signal IE in the ALERTING, CONNECT, and RELEASE messages and the telco complains that the IE is not supported.
>  
>  
>  
> RX [08 02 32 11 05 a1 04 03 80 90 a2 18 03 a1 83 97 34 01 40 6c 0a 00 81 XX XX XX XX 30 36 39 34 70 05 80 36 37 30 30]
> Message Type: [05] SETUP
> IE: [a1] Sending Complete
> IE: [04] Bearer Capability
> IE: [18] Channel identification
> IE: [34] Signal - Alerting on ? pattern 0
> IE: [6c] Calling party number - TON unknown/NPI unknown/Presentation allowed/User-provided, not screened/XXXX0694
> IE: [70] Called party number - TON unknown/NPI unknown/6700
>  
>  
> RX [08 02 32 11 05 a1 04 03 80 90 a2 18 03 a1 83 97 34 01 40 6c 0a 00 81 XX XX XX XX 30 36 39 34 70 05 80 36 37 30 30]
> Message Type: [05] SETUP
> IE: [a1] Sending Complete
> IE: [04] Bearer Capability
> IE: [18] Channel identification
> IE: [34] Signal - Alerting on ? pattern 0
> IE: [6c] Calling party number - TON unknown/NPI unknown/Presentation allowed/User-provided, not screened/XXXX0694
> IE: [70] Called party number - TON unknown/NPI unknown/6700
>  
>  
> TX [08 02 b2 11 02 04 03 80 90 a2 18 03 a1 83 97]
> Message Type: [02] CALL PROCEEDING
> IE: [04] Bearer Capability
> IE: [18] Channel identification
>  
>  
> TX [08 02 b2 11 01 04 03 80 90 a2 18 03 a1 83 97 34 01 40]
> Message Type: [01] ALERTING
> IE: [04] Bearer Capability
> IE: [18] Channel identification
> IE: [34] Signal - Alerting on ? pattern 0
>  
>  
> TX [08 02 b2 11 07 18 03 a1 83 97 34 01 40]
> Message Type: [07] CONNECT
> IE: [18] Channel identification
> IE: [34] Signal - Alerting on ? pattern 0
>  
>  
> RX [08 02 32 11 7d 08 07 82 e3 34 00 00 00 00 14 01 07]
> Message Type: [7d] STATUS
> IE: [08] Cause -- IE not implemented/Signal
> IE: [14] Call State -- call received
>  
>  
> RX [08 02 32 11 7d 08 07 82 e3 34 00 00 00 00 14 01 08]
> Message Type: [7d] STATUS
> IE: [08] Cause -- IE not implemented/Signal
> IE: [14] Call State -- connect request
>  
>  
> RX [08 02 32 11 0f]
> Message Type: [0f] CONNECT ACKNOWLEDGE
>  
>  
> RX [08 02 32 11 45 08 02 82 90]
> Message Type: [45] DISCONNECT
> IE: [08] Cause -- Normal call clearing
>  
>  
> TX [08 02 b2 11 4d 34 01 40]
> Message Type: [4d] RELEASE
> IE: [34] Signal - Alerting on ? pattern 0
>  
>  
> RX [08 02 32 11 5a 08 07 82 e3 34 00 00 00 00]
> Message Type: [5a] RELEASE COMPLETE
> IE: [08] Cause -- IE not implemented/Signal
>  
>  
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale
> Sent: Monday, May 10, 2010 9:15 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] short ringback tone when answer
>  
> if ringback is not set, it would never play a tone and even if it was, it would not be played if you explicitly called answer.
> Try reversing the answer and the sleep in your dialplan.
>  
>  
>  
> 
> On Sun, May 9, 2010 at 9:37 PM, Tony Tin <tony.tin at noahmedia.com.hk> wrote:
> Hi Peter,
> 
> Thanks for your reply.
> 
> The other end is the Telco, I've confirmed with them that they don't play the tone for us, also I got the Asterisk install in the same box, it doesn't play the tone. The tone is played exactly when the answer function is executed, if I put sleep 10 seconds before the answer, the tone will delay 10 seconds too, any idea?
> 
> Regards,
> Tony
> 
> 
> 
> 
> On Mon, May 10, 2010 at 2:33 AM, Peter Olsson <peter.olsson at visionutveckling.se> wrote:
> The equipment connected in the other en of the T1 is probably playing these tones for you, 500ms is probably the time it takes for the called to be connected successfully.
> 
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Tony Tin [tony.tin at noahmedia.com.hk]
> Skickat: den 9 maj 2010 15:16
> Till: freeswitch-users at lists.freeswitch.org
> ?mne: [Freeswitch-users] short ringback tone when answer
> 
> Hi All,
> 
> I've just setup a new IVRS with FreeSwitch and a Digium TE220 card (which is connected to a 4ess T1). Every thing works fine so far except one little problem that annoys me. Every time I dial in the ivrs, there is a very short (0.5 second) ringback tone exactly while the "<action application="answer"/>" line is executed in the dial plan. I want to disable this ringback tone because the ivrs will answer the call immediately, the ringback tone is really unnecessary in this case, but I fail to do that. I try to set the variable "ringback" to change the ringback tone, it's also not working. Could anyone please help.
> 
> I'm using the native pri stack.
> The default.xml containts only below lines, and attached is the log file.
> 
>       <extension name="testing">
>          <condition field="destination_number" expression="^(6.*)$">
>                <action application="sleep" data="3000"/>
>                <action application="answer"/>
>                <action application="sleep" data="3000"/>
>          </condition>
>       </extension>
> 
> Regards,
> Tony
> 
> !DSPAM:4be6f37432931620317181!
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>  
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jan.berger at video24.no  Tue May 11 16:18:18 2010
From: jan.berger at video24.no (Jan Berger)
Date: Wed, 12 May 2010 01:18:18 +0200
Subject: [Freeswitch-users] short ringback tone when answer
In-Reply-To: <6265B4F0F9F949759F3154920D7F4416@fromage>
References: <j2g191e3b11005090616ndd0b0ef8g37f60c1f1016ab4f@mail.gmail.com><549CFEF87AEDE841A38E9D15EAB4C04C5576CE4DC7@cooper><y2w191e3b11005091937k6ea8d442i798b95555e11e817@mail.gmail.com><AANLkTikFk61Phm9lKYIh33BfsMFPfgFZrO-rivpJzi3v@mail.gmail.com><D0216DB842C84F77999189E6FC9CF38E@fromage>
	<B04A76A2-A198-4E01-95FD-A4BBCF8ED4D5@jerris.com>
	<6265B4F0F9F949759F3154920D7F4416@fromage>
Message-ID: <1033D8D145DA498C8717210040F0BAC1@dell9400>

Ron,

 

What equipment and what Q.931 variant are you connected to? If possible get
a PICS out of them.

 

L3 should respond with SETUPACK on a SETUP to avoid re-send. You can respond
with CallProceeding or Alerting, but SetupAck is designed to be sent back
from L3 to buy L4 time.

 

Also ? you need to disable sending of Signal. It is not a mandatory part of
the stack, but I would like to know what equipment/ISDN variant we are
daling with here.

 

Also ? is this OpenZAP, Libpri or a sangoma stack you use on FreeSWITCH ?

 

Jan

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ron
McLeod
Sent: 11. mai 2010 23:30
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

FreeSWITCH ISDN.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Jerris
Sent: Tuesday, May 11, 2010 11:29 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

which pri stack is this?

 

Mike

 

On May 11, 2010, at 1:35 PM, Ron McLeod wrote:

 

Tony sent me a Q.931 trace off-list and I thought I would post it here any
case anyone has seen this before.

 

The telco side is including a Signal IE in the SETUP message.  I?m not sure
what this means ? that the telco will play alerting pattern 0 back to the
CPE unless otherwise instructed?

 

Other things in the trace:

  * SETUP is sent again because the dial plan was executing sleep() before
anything triggered a CALL PROCEEDING.

  * The switch PRI stack is reflecting back the Signal IE in the ALERTING,
CONNECT, and RELEASE messages and the telco complains that the IE is not
supported.

 

 

 

RX [08 02 32 11 05 a1 04 03 80 90 a2 18 03 a1 83 97 34 01 40 6c 0a 00 81 XX
XX XX XX 30 36 39 34 70 05 80 36 37 30 30]

Message Type: [05] SETUP

IE: [a1] Sending Complete

IE: [04] Bearer Capability

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

IE: [6c] Calling party number - TON unknown/NPI unknown/Presentation
allowed/User-provided, not screened/XXXX0694

IE: [70] Called party number - TON unknown/NPI unknown/6700

 

 

RX [08 02 32 11 05 a1 04 03 80 90 a2 18 03 a1 83 97 34 01 40 6c 0a 00 81 XX
XX XX XX 30 36 39 34 70 05 80 36 37 30 30]

Message Type: [05] SETUP

IE: [a1] Sending Complete

IE: [04] Bearer Capability

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

IE: [6c] Calling party number - TON unknown/NPI unknown/Presentation
allowed/User-provided, not screened/XXXX0694

IE: [70] Called party number - TON unknown/NPI unknown/6700

 

 

TX [08 02 b2 11 02 04 03 80 90 a2 18 03 a1 83 97]

Message Type: [02] CALL PROCEEDING

IE: [04] Bearer Capability

IE: [18] Channel identification

 

 

TX [08 02 b2 11 01 04 03 80 90 a2 18 03 a1 83 97 34 01 40]

Message Type: [01] ALERTING

IE: [04] Bearer Capability

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

 

 

TX [08 02 b2 11 07 18 03 a1 83 97 34 01 40]

Message Type: [07] CONNECT

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

 

 

RX [08 02 32 11 7d 08 07 82 e3 34 00 00 00 00 14 01 07]

Message Type: [7d] STATUS

IE: [08] Cause -- IE not implemented/Signal

IE: [14] Call State -- call received

 

 

RX [08 02 32 11 7d 08 07 82 e3 34 00 00 00 00 14 01 08]

Message Type: [7d] STATUS

IE: [08] Cause -- IE not implemented/Signal

IE: [14] Call State -- connect request

 

 

RX [08 02 32 11 0f]

Message Type: [0f] CONNECT ACKNOWLEDGE

 

 

RX [08 02 32 11 45 08 02 82 90]

Message Type: [45] DISCONNECT

IE: [08] Cause -- Normal call clearing

 

 

TX [08 02 b2 11 4d 34 01 40]

Message Type: [4d] RELEASE

IE: [34] Signal - Alerting on ? pattern 0

 

 

RX [08 02 32 11 5a 08 07 82 e3 34 00 00 00 00]

Message Type: [5a] RELEASE COMPLETE

IE: [08] Cause -- IE not implemented/Signal

 

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, May 10, 2010 9:15 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

if ringback is not set, it would never play a tone and even if it was, it
would not be played if you explicitly called answer.

Try reversing the answer and the sleep in your dialplan.

 

 

 

On Sun, May 9, 2010 at 9:37 PM, Tony Tin <tony.tin at noahmedia.com.hk> wrote:

Hi Peter,

Thanks for your reply.

The other end is the Telco, I've confirmed with them that they don't play
the tone for us, also I got the Asterisk install in the same box, it doesn't
play the tone. The tone is played exactly when the answer function is
executed, if I put sleep 10 seconds before the answer, the tone will delay
10 seconds too, any idea?

Regards,
Tony





On Mon, May 10, 2010 at 2:33 AM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:

The equipment connected in the other en of the T1 is probably playing these
tones for you, 500ms is probably the time it takes for the called to be
connected successfully.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org
[freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Tony Tin
[tony.tin at noahmedia.com.hk]
Skickat: den 9 maj 2010 15:16
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] short ringback tone when answer


Hi All,

I've just setup a new IVRS with FreeSwitch and a Digium TE220 card (which is
connected to a 4ess T1). Every thing works fine so far except one little
problem that annoys me. Every time I dial in the ivrs, there is a very short
(0.5 second) ringback tone exactly while the "<action
application="answer"/>" line is executed in the dial plan. I want to disable
this ringback tone because the ivrs will answer the call immediately, the
ringback tone is really unnecessary in this case, but I fail to do that. I
try to set the variable "ringback" to change the ringback tone, it's also
not working. Could anyone please help.

I'm using the native pri stack.
The default.xml containts only below lines, and attached is the log file.

      <extension name="testing">
         <condition field="destination_number" expression="^(6.*)$">
               <action application="sleep" data="3000"/>
               <action application="answer"/>
               <action application="sleep" data="3000"/>
         </condition>
      </extension>

Regards,
Tony

!DSPAM:4be6f37432931620317181!

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

 


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




-- 
Anthony Minessale II

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

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

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

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

 

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From anthony.minessale at gmail.com  Tue May 11 16:27:49 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 11 May 2010 18:27:49 -0500
Subject: [Freeswitch-users] Freeswitch Scripting strangeness
In-Reply-To: <4BE9CE90.7050805@conspiracy.net>
References: <4BE9CE90.7050805@conspiracy.net>
Message-ID: <AANLkTikpnpX4YxezY7BpkrI0-eMNC5JuXXhZQAMysWHt@mail.gmail.com>

Your assumption is correct.

however,

open file handles are not threadsafe and you cannot have 2 threads opening
the same file and expect sane results.  That's why syslog is a udp network
server so it can serialize the log messages into a single threaded open file
handle in the central syslog process.


On Tue, May 11, 2010 at 4:39 PM, paul bilke <fsmail at conspiracy.net> wrote:

> I am working on a system that invokes a LUA script of the dial plan to
> handle interaction on the call.
> All is fine when running a single instance of the script (one call).  The
> issue is when there is more than one call things become erratic.
> Right now the script opens a log file who's name is based on the date of
> the call and its DNIS using io.open with append.
> The script then writes lines to file as it runs detailing its actions.  The
> same log function also uses posix.syslog to log the same information
> via syslog.  All is fine when there is one call up.
> If there is more than one concurrent on two different DNIS's both log files
> a present but have output that was created by the other call.
> The syslog is sane but not the file of logging information.  The time stamp
> will jump backward and pickup with information from the other call
> (can tell via the UUID)
> It is my assumption that each call should spin up a thread and they should
> share no data/variables/handles with other calls in the system.  Looking at
> the logs indicates there is some issue but logging is not the only issue
> since it seems that call events updating state information is affecting the
> other
> call also.
> If my assumption on a isolated thread per call is incorrect let me know.
> I inherited this code and suspect that running an application out of the
> dialplan script is unconventional.
> Paul
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <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
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From ron.freeswitch at mcleodnet.com  Tue May 11 16:58:36 2010
From: ron.freeswitch at mcleodnet.com (Ron McLeod)
Date: Tue, 11 May 2010 16:58:36 -0700
Subject: [Freeswitch-users] short ringback tone when answer
In-Reply-To: <1033D8D145DA498C8717210040F0BAC1@dell9400>
References: <j2g191e3b11005090616ndd0b0ef8g37f60c1f1016ab4f@mail.gmail.com><549CFEF87AEDE841A38E9D15EAB4C04C5576CE4DC7@cooper><y2w191e3b11005091937k6ea8d442i798b95555e11e817@mail.gmail.com><AANLkTikFk61Phm9lKYIh33BfsMFPfgFZrO-rivpJzi3v@mail.gmail.com><D0216DB842C84F77999189E6FC9CF38E@fromage><B04A76A2-A198-4E01-95FD-A4BBCF8ED4D5@jerris.com><6265B4F0F9F949759F3154920D7F4416@fromage>
	<1033D8D145DA498C8717210040F0BAC1@dell9400>
Message-ID: <3F7EDFD0AB0C46C7880364C2FA9DA27A@fromage>

Jan ?

 

This is not my issue ? I am helping someone in HKG that posted earlier
today, but I can answer some the questions on his behalf.

 

I?m not sure of the make/model of the PSTN switch (he may not know either),
but the interface is T1 and the variant is 4ESS.

 

FreeSWITCH is the terminating side of the call in his application, and is
configured as the Q.931 user.  It does send CALL PROCEEDING message in
response to the SETUP message, but it is delayed because he had a 6 second
sleep as the first action in dial plan and PSTN side timed-out waiting (T303
= 4secs).  The stack should probably send a SETUP ACKNOWLEDGE after it finds
a match in the dial plan, before any actions are executed.

 

The PSTN is the one sending the Signal IE in the SETUP message, not
FreeSWITCH.

 

The stack is FreeSWITCH ISDN (not Sangoma, not libpri).

 

What I would like to understand is -- what does including a Signal IE in the
SETUP message actually mean.  Does it inform the terminating end that the
originating end will play the specified call progression to the calling
party automatically (this appears to be what is happening)?  Is it a request
to the terminating switch of what type of in-band call information it should
send back to the calling party?

 

Ron

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jan
Berger
Sent: Tuesday, May 11, 2010 4:18 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

Ron,

 

What equipment and what Q.931 variant are you connected to? If possible get
a PICS out of them.

 

L3 should respond with SETUPACK on a SETUP to avoid re-send. You can respond
with CallProceeding or Alerting, but SetupAck is designed to be sent back
from L3 to buy L4 time.

 

Also ? you need to disable sending of Signal. It is not a mandatory part of
the stack, but I would like to know what equipment/ISDN variant we are
daling with here.

 

Also ? is this OpenZAP, Libpri or a sangoma stack you use on FreeSWITCH ?

 

Jan

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ron
McLeod
Sent: 11. mai 2010 23:30
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

FreeSWITCH ISDN.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Jerris
Sent: Tuesday, May 11, 2010 11:29 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

which pri stack is this?

 

Mike

 

On May 11, 2010, at 1:35 PM, Ron McLeod wrote:

 

Tony sent me a Q.931 trace off-list and I thought I would post it here any
case anyone has seen this before.

 

The telco side is including a Signal IE in the SETUP message.  I?m not sure
what this means ? that the telco will play alerting pattern 0 back to the
CPE unless otherwise instructed?

 

Other things in the trace:

  * SETUP is sent again because the dial plan was executing sleep() before
anything triggered a CALL PROCEEDING.

  * The switch PRI stack is reflecting back the Signal IE in the ALERTING,
CONNECT, and RELEASE messages and the telco complains that the IE is not
supported.

 

 

 

RX [08 02 32 11 05 a1 04 03 80 90 a2 18 03 a1 83 97 34 01 40 6c 0a 00 81 XX
XX XX XX 30 36 39 34 70 05 80 36 37 30 30]

Message Type: [05] SETUP

IE: [a1] Sending Complete

IE: [04] Bearer Capability

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

IE: [6c] Calling party number - TON unknown/NPI unknown/Presentation
allowed/User-provided, not screened/XXXX0694

IE: [70] Called party number - TON unknown/NPI unknown/6700

 

 

RX [08 02 32 11 05 a1 04 03 80 90 a2 18 03 a1 83 97 34 01 40 6c 0a 00 81 XX
XX XX XX 30 36 39 34 70 05 80 36 37 30 30]

Message Type: [05] SETUP

IE: [a1] Sending Complete

IE: [04] Bearer Capability

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

IE: [6c] Calling party number - TON unknown/NPI unknown/Presentation
allowed/User-provided, not screened/XXXX0694

IE: [70] Called party number - TON unknown/NPI unknown/6700

 

 

TX [08 02 b2 11 02 04 03 80 90 a2 18 03 a1 83 97]

Message Type: [02] CALL PROCEEDING

IE: [04] Bearer Capability

IE: [18] Channel identification

 

 

TX [08 02 b2 11 01 04 03 80 90 a2 18 03 a1 83 97 34 01 40]

Message Type: [01] ALERTING

IE: [04] Bearer Capability

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

 

 

TX [08 02 b2 11 07 18 03 a1 83 97 34 01 40]

Message Type: [07] CONNECT

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

 

 

RX [08 02 32 11 7d 08 07 82 e3 34 00 00 00 00 14 01 07]

Message Type: [7d] STATUS

IE: [08] Cause -- IE not implemented/Signal

IE: [14] Call State -- call received

 

 

RX [08 02 32 11 7d 08 07 82 e3 34 00 00 00 00 14 01 08]

Message Type: [7d] STATUS

IE: [08] Cause -- IE not implemented/Signal

IE: [14] Call State -- connect request

 

 

RX [08 02 32 11 0f]

Message Type: [0f] CONNECT ACKNOWLEDGE

 

 

RX [08 02 32 11 45 08 02 82 90]

Message Type: [45] DISCONNECT

IE: [08] Cause -- Normal call clearing

 

 

TX [08 02 b2 11 4d 34 01 40]

Message Type: [4d] RELEASE

IE: [34] Signal - Alerting on ? pattern 0

 

 

RX [08 02 32 11 5a 08 07 82 e3 34 00 00 00 00]

Message Type: [5a] RELEASE COMPLETE

IE: [08] Cause -- IE not implemented/Signal

 

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, May 10, 2010 9:15 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

if ringback is not set, it would never play a tone and even if it was, it
would not be played if you explicitly called answer.

Try reversing the answer and the sleep in your dialplan.

 

 

 

On Sun, May 9, 2010 at 9:37 PM, Tony Tin <tony.tin at noahmedia.com.hk> wrote:

Hi Peter,

Thanks for your reply.

The other end is the Telco, I've confirmed with them that they don't play
the tone for us, also I got the Asterisk install in the same box, it doesn't
play the tone. The tone is played exactly when the answer function is
executed, if I put sleep 10 seconds before the answer, the tone will delay
10 seconds too, any idea?

Regards,
Tony

 

On Mon, May 10, 2010 at 2:33 AM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:

The equipment connected in the other en of the T1 is probably playing these
tones for you, 500ms is probably the time it takes for the called to be
connected successfully.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org
[freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Tony Tin
[tony.tin at noahmedia.com.hk]
Skickat: den 9 maj 2010 15:16
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] short ringback tone when answer


Hi All,

I've just setup a new IVRS with FreeSwitch and a Digium TE220 card (which is
connected to a 4ess T1). Every thing works fine so far except one little
problem that annoys me. Every time I dial in the ivrs, there is a very short
(0.5 second) ringback tone exactly while the "<action
application="answer"/>" line is executed in the dial plan. I want to disable
this ringback tone because the ivrs will answer the call immediately, the
ringback tone is really unnecessary in this case, but I fail to do that. I
try to set the variable "ringback" to change the ringback tone, it's also
not working. Could anyone please help.

I'm using the native pri stack.
The default.xml containts only below lines, and attached is the log file.

      <extension name="testing">
         <condition field="destination_number" expression="^(6.*)$">
               <action application="sleep" data="3000"/>
               <action application="answer"/>
               <action application="sleep" data="3000"/>
         </condition>
      </extension>

Regards,
Tony

!DSPAM:4be6f37432931620317181!

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

 


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




-- 
Anthony Minessale II

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

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

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

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

 

-------------- next part --------------
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From jan.berger at video24.no  Tue May 11 17:28:36 2010
From: jan.berger at video24.no (Jan Berger)
Date: Wed, 12 May 2010 02:28:36 +0200
Subject: [Freeswitch-users] short ringback tone when answer
In-Reply-To: <3F7EDFD0AB0C46C7880364C2FA9DA27A@fromage>
References: <j2g191e3b11005090616ndd0b0ef8g37f60c1f1016ab4f@mail.gmail.com><549CFEF87AEDE841A38E9D15EAB4C04C5576CE4DC7@cooper><y2w191e3b11005091937k6ea8d442i798b95555e11e817@mail.gmail.com><AANLkTikFk61Phm9lKYIh33BfsMFPfgFZrO-rivpJzi3v@mail.gmail.com><D0216DB842C84F77999189E6FC9CF38E@fromage><B04A76A2-A198-4E01-95FD-A4BBCF8ED4D5@jerris.com><6265B4F0F9F949759F3154920D7F4416@fromage>
	<1033D8D145DA498C8717210040F0BAC1@dell9400>
	<3F7EDFD0AB0C46C7880364C2FA9DA27A@fromage>
Message-ID: <F2F9159C1CD04D4DB1E72114507EDCE7@dell9400>

4ESS means the make is AT&T ? it?s an old legacy thing ? ISDN PRI only. You
can get the standard doc from AT&T for free ? at least it used to be on
their site.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ron
McLeod
Sent: 12. mai 2010 01:59
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

Jan ?

 

This is not my issue ? I am helping someone in HKG that posted earlier
today, but I can answer some the questions on his behalf.

 

I?m not sure of the make/model of the PSTN switch (he may not know either),
but the interface is T1 and the variant is 4ESS.

 

FreeSWITCH is the terminating side of the call in his application, and is
configured as the Q.931 user.  It does send CALL PROCEEDING message in
response to the SETUP message, but it is delayed because he had a 6 second
sleep as the first action in dial plan and PSTN side timed-out waiting (T303
= 4secs).  The stack should probably send a SETUP ACKNOWLEDGE after it finds
a match in the dial plan, before any actions are executed.

 

The PSTN is the one sending the Signal IE in the SETUP message, not
FreeSWITCH.

 

The stack is FreeSWITCH ISDN (not Sangoma, not libpri).

 

What I would like to understand is -- what does including a Signal IE in the
SETUP message actually mean.  Does it inform the terminating end that the
originating end will play the specified call progression to the calling
party automatically (this appears to be what is happening)?  Is it a request
to the terminating switch of what type of in-band call information it should
send back to the calling party?

 

Ron

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jan
Berger
Sent: Tuesday, May 11, 2010 4:18 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

Ron,

 

What equipment and what Q.931 variant are you connected to? If possible get
a PICS out of them.

 

L3 should respond with SETUPACK on a SETUP to avoid re-send. You can respond
with CallProceeding or Alerting, but SetupAck is designed to be sent back
from L3 to buy L4 time.

 

Also ? you need to disable sending of Signal. It is not a mandatory part of
the stack, but I would like to know what equipment/ISDN variant we are
daling with here.

 

Also ? is this OpenZAP, Libpri or a sangoma stack you use on FreeSWITCH ?

 

Jan

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ron
McLeod
Sent: 11. mai 2010 23:30
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

FreeSWITCH ISDN.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Jerris
Sent: Tuesday, May 11, 2010 11:29 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

which pri stack is this?

 

Mike

 

On May 11, 2010, at 1:35 PM, Ron McLeod wrote:

 

Tony sent me a Q.931 trace off-list and I thought I would post it here any
case anyone has seen this before.

 

The telco side is including a Signal IE in the SETUP message.  I?m not sure
what this means ? that the telco will play alerting pattern 0 back to the
CPE unless otherwise instructed?

 

Other things in the trace:

  * SETUP is sent again because the dial plan was executing sleep() before
anything triggered a CALL PROCEEDING.

  * The switch PRI stack is reflecting back the Signal IE in the ALERTING,
CONNECT, and RELEASE messages and the telco complains that the IE is not
supported.

 

 

 

RX [08 02 32 11 05 a1 04 03 80 90 a2 18 03 a1 83 97 34 01 40 6c 0a 00 81 XX
XX XX XX 30 36 39 34 70 05 80 36 37 30 30]

Message Type: [05] SETUP

IE: [a1] Sending Complete

IE: [04] Bearer Capability

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

IE: [6c] Calling party number - TON unknown/NPI unknown/Presentation
allowed/User-provided, not screened/XXXX0694

IE: [70] Called party number - TON unknown/NPI unknown/6700

 

 

RX [08 02 32 11 05 a1 04 03 80 90 a2 18 03 a1 83 97 34 01 40 6c 0a 00 81 XX
XX XX XX 30 36 39 34 70 05 80 36 37 30 30]

Message Type: [05] SETUP

IE: [a1] Sending Complete

IE: [04] Bearer Capability

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

IE: [6c] Calling party number - TON unknown/NPI unknown/Presentation
allowed/User-provided, not screened/XXXX0694

IE: [70] Called party number - TON unknown/NPI unknown/6700

 

 

TX [08 02 b2 11 02 04 03 80 90 a2 18 03 a1 83 97]

Message Type: [02] CALL PROCEEDING

IE: [04] Bearer Capability

IE: [18] Channel identification

 

 

TX [08 02 b2 11 01 04 03 80 90 a2 18 03 a1 83 97 34 01 40]

Message Type: [01] ALERTING

IE: [04] Bearer Capability

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

 

 

TX [08 02 b2 11 07 18 03 a1 83 97 34 01 40]

Message Type: [07] CONNECT

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

 

 

RX [08 02 32 11 7d 08 07 82 e3 34 00 00 00 00 14 01 07]

Message Type: [7d] STATUS

IE: [08] Cause -- IE not implemented/Signal

IE: [14] Call State -- call received

 

 

RX [08 02 32 11 7d 08 07 82 e3 34 00 00 00 00 14 01 08]

Message Type: [7d] STATUS

IE: [08] Cause -- IE not implemented/Signal

IE: [14] Call State -- connect request

 

 

RX [08 02 32 11 0f]

Message Type: [0f] CONNECT ACKNOWLEDGE

 

 

RX [08 02 32 11 45 08 02 82 90]

Message Type: [45] DISCONNECT

IE: [08] Cause -- Normal call clearing

 

 

TX [08 02 b2 11 4d 34 01 40]

Message Type: [4d] RELEASE

IE: [34] Signal - Alerting on ? pattern 0

 

 

RX [08 02 32 11 5a 08 07 82 e3 34 00 00 00 00]

Message Type: [5a] RELEASE COMPLETE

IE: [08] Cause -- IE not implemented/Signal

 

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, May 10, 2010 9:15 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

if ringback is not set, it would never play a tone and even if it was, it
would not be played if you explicitly called answer.

Try reversing the answer and the sleep in your dialplan.

 

 

 

On Sun, May 9, 2010 at 9:37 PM, Tony Tin <tony.tin at noahmedia.com.hk> wrote:

Hi Peter,

Thanks for your reply.

The other end is the Telco, I've confirmed with them that they don't play
the tone for us, also I got the Asterisk install in the same box, it doesn't
play the tone. The tone is played exactly when the answer function is
executed, if I put sleep 10 seconds before the answer, the tone will delay
10 seconds too, any idea?

Regards,
Tony

 

On Mon, May 10, 2010 at 2:33 AM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:

The equipment connected in the other en of the T1 is probably playing these
tones for you, 500ms is probably the time it takes for the called to be
connected successfully.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org
[freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Tony Tin
[tony.tin at noahmedia.com.hk]
Skickat: den 9 maj 2010 15:16
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] short ringback tone when answer


Hi All,

I've just setup a new IVRS with FreeSwitch and a Digium TE220 card (which is
connected to a 4ess T1). Every thing works fine so far except one little
problem that annoys me. Every time I dial in the ivrs, there is a very short
(0.5 second) ringback tone exactly while the "<action
application="answer"/>" line is executed in the dial plan. I want to disable
this ringback tone because the ivrs will answer the call immediately, the
ringback tone is really unnecessary in this case, but I fail to do that. I
try to set the variable "ringback" to change the ringback tone, it's also
not working. Could anyone please help.

I'm using the native pri stack.
The default.xml containts only below lines, and attached is the log file.

      <extension name="testing">
         <condition field="destination_number" expression="^(6.*)$">
               <action application="sleep" data="3000"/>
               <action application="answer"/>
               <action application="sleep" data="3000"/>
         </condition>
      </extension>

Regards,
Tony

!DSPAM:4be6f37432931620317181!

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

 


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




-- 
Anthony Minessale II

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

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

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

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

 

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From lntasin at gmail.com  Tue May 11 17:16:46 2010
From: lntasin at gmail.com (Louis Ntasin)
Date: Tue, 11 May 2010 20:16:46 -0400
Subject: [Freeswitch-users] Mod_lcr question...
Message-ID: <AANLkTikL3yFesdq9xkxmYnx2_SD5kTPclUakOHOrj5rE@mail.gmail.com>

Gents-

I am new to this list and would like to say hi to all. I have a question
about mod_lcr.

When I load mod_lcr I see errors about a missing table: npa_nxx_company_ocn.
I did some searching and saw that it is meant for intra lata/state calling.
I don't need this feature. Is there a flag that I can turn on to tell
mod_lcr not to worry about the table?

Thanks.

Louis
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From ron.freeswitch at mcleodnet.com  Tue May 11 17:47:57 2010
From: ron.freeswitch at mcleodnet.com (Ron McLeod)
Date: Tue, 11 May 2010 17:47:57 -0700
Subject: [Freeswitch-users] short ringback tone when answer
In-Reply-To: <F2F9159C1CD04D4DB1E72114507EDCE7@dell9400>
References: <j2g191e3b11005090616ndd0b0ef8g37f60c1f1016ab4f@mail.gmail.com><549CFEF87AEDE841A38E9D15EAB4C04C5576CE4DC7@cooper><y2w191e3b11005091937k6ea8d442i798b95555e11e817@mail.gmail.com><AANLkTikFk61Phm9lKYIh33BfsMFPfgFZrO-rivpJzi3v@mail.gmail.com><D0216DB842C84F77999189E6FC9CF38E@fromage><B04A76A2-A198-4E01-95FD-A4BBCF8ED4D5@jerris.com><6265B4F0F9F949759F3154920D7F4416@fromage><1033D8D145DA498C8717210040F0BAC1@dell9400><3F7EDFD0AB0C46C7880364C2FA9DA27A@fromage>
	<F2F9159C1CD04D4DB1E72114507EDCE7@dell9400>
Message-ID: <867B555E693B4B71AC3D81EDD57DC750@fromage>

I have the docs ? they are pretty much the same as the ITU docs except for
some additions for propagating traveling class marks and other proprietary
stuff.  I don?t think this issue is variant-specific anyway.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jan
Berger
Sent: Tuesday, May 11, 2010 5:29 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

4ESS means the make is AT&T ? it?s an old legacy thing ? ISDN PRI only. You
can get the standard doc from AT&T for free ? at least it used to be on
their site.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ron
McLeod
Sent: 12. mai 2010 01:59
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

Jan ?

 

This is not my issue ? I am helping someone in HKG that posted earlier
today, but I can answer some the questions on his behalf.

 

I?m not sure of the make/model of the PSTN switch (he may not know either),
but the interface is T1 and the variant is 4ESS.

 

FreeSWITCH is the terminating side of the call in his application, and is
configured as the Q.931 user.  It does send CALL PROCEEDING message in
response to the SETUP message, but it is delayed because he had a 6 second
sleep as the first action in dial plan and PSTN side timed-out waiting (T303
= 4secs).  The stack should probably send a SETUP ACKNOWLEDGE after it finds
a match in the dial plan, before any actions are executed.

 

The PSTN is the one sending the Signal IE in the SETUP message, not
FreeSWITCH.

 

The stack is FreeSWITCH ISDN (not Sangoma, not libpri).

 

What I would like to understand is -- what does including a Signal IE in the
SETUP message actually mean.  Does it inform the terminating end that the
originating end will play the specified call progression to the calling
party automatically (this appears to be what is happening)?  Is it a request
to the terminating switch of what type of in-band call information it should
send back to the calling party?

 

Ron

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jan
Berger
Sent: Tuesday, May 11, 2010 4:18 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

Ron,

 

What equipment and what Q.931 variant are you connected to? If possible get
a PICS out of them.

 

L3 should respond with SETUPACK on a SETUP to avoid re-send. You can respond
with CallProceeding or Alerting, but SetupAck is designed to be sent back
from L3 to buy L4 time.

 

Also ? you need to disable sending of Signal. It is not a mandatory part of
the stack, but I would like to know what equipment/ISDN variant we are
daling with here.

 

Also ? is this OpenZAP, Libpri or a sangoma stack you use on FreeSWITCH ?

 

Jan

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ron
McLeod
Sent: 11. mai 2010 23:30
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

FreeSWITCH ISDN.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Jerris
Sent: Tuesday, May 11, 2010 11:29 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

which pri stack is this?

 

Mike

 

On May 11, 2010, at 1:35 PM, Ron McLeod wrote:

 

Tony sent me a Q.931 trace off-list and I thought I would post it here any
case anyone has seen this before.

 

The telco side is including a Signal IE in the SETUP message.  I?m not sure
what this means ? that the telco will play alerting pattern 0 back to the
CPE unless otherwise instructed?

 

Other things in the trace:

  * SETUP is sent again because the dial plan was executing sleep() before
anything triggered a CALL PROCEEDING.

  * The switch PRI stack is reflecting back the Signal IE in the ALERTING,
CONNECT, and RELEASE messages and the telco complains that the IE is not
supported.

 

 

 

RX [08 02 32 11 05 a1 04 03 80 90 a2 18 03 a1 83 97 34 01 40 6c 0a 00 81 XX
XX XX XX 30 36 39 34 70 05 80 36 37 30 30]

Message Type: [05] SETUP

IE: [a1] Sending Complete

IE: [04] Bearer Capability

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

IE: [6c] Calling party number - TON unknown/NPI unknown/Presentation
allowed/User-provided, not screened/XXXX0694

IE: [70] Called party number - TON unknown/NPI unknown/6700

 

 

RX [08 02 32 11 05 a1 04 03 80 90 a2 18 03 a1 83 97 34 01 40 6c 0a 00 81 XX
XX XX XX 30 36 39 34 70 05 80 36 37 30 30]

Message Type: [05] SETUP

IE: [a1] Sending Complete

IE: [04] Bearer Capability

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

IE: [6c] Calling party number - TON unknown/NPI unknown/Presentation
allowed/User-provided, not screened/XXXX0694

IE: [70] Called party number - TON unknown/NPI unknown/6700

 

 

TX [08 02 b2 11 02 04 03 80 90 a2 18 03 a1 83 97]

Message Type: [02] CALL PROCEEDING

IE: [04] Bearer Capability

IE: [18] Channel identification

 

 

TX [08 02 b2 11 01 04 03 80 90 a2 18 03 a1 83 97 34 01 40]

Message Type: [01] ALERTING

IE: [04] Bearer Capability

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

 

 

TX [08 02 b2 11 07 18 03 a1 83 97 34 01 40]

Message Type: [07] CONNECT

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

 

 

RX [08 02 32 11 7d 08 07 82 e3 34 00 00 00 00 14 01 07]

Message Type: [7d] STATUS

IE: [08] Cause -- IE not implemented/Signal

IE: [14] Call State -- call received

 

 

RX [08 02 32 11 7d 08 07 82 e3 34 00 00 00 00 14 01 08]

Message Type: [7d] STATUS

IE: [08] Cause -- IE not implemented/Signal

IE: [14] Call State -- connect request

 

 

RX [08 02 32 11 0f]

Message Type: [0f] CONNECT ACKNOWLEDGE

 

 

RX [08 02 32 11 45 08 02 82 90]

Message Type: [45] DISCONNECT

IE: [08] Cause -- Normal call clearing

 

 

TX [08 02 b2 11 4d 34 01 40]

Message Type: [4d] RELEASE

IE: [34] Signal - Alerting on ? pattern 0

 

 

RX [08 02 32 11 5a 08 07 82 e3 34 00 00 00 00]

Message Type: [5a] RELEASE COMPLETE

IE: [08] Cause -- IE not implemented/Signal

 

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, May 10, 2010 9:15 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

if ringback is not set, it would never play a tone and even if it was, it
would not be played if you explicitly called answer.

Try reversing the answer and the sleep in your dialplan.

 

 

 

On Sun, May 9, 2010 at 9:37 PM, Tony Tin <tony.tin at noahmedia.com.hk> wrote:

Hi Peter,

Thanks for your reply.

The other end is the Telco, I've confirmed with them that they don't play
the tone for us, also I got the Asterisk install in the same box, it doesn't
play the tone. The tone is played exactly when the answer function is
executed, if I put sleep 10 seconds before the answer, the tone will delay
10 seconds too, any idea?

Regards,
Tony

 

On Mon, May 10, 2010 at 2:33 AM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:

The equipment connected in the other en of the T1 is probably playing these
tones for you, 500ms is probably the time it takes for the called to be
connected successfully.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org
[freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Tony Tin
[tony.tin at noahmedia.com.hk]
Skickat: den 9 maj 2010 15:16
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] short ringback tone when answer


Hi All,

I've just setup a new IVRS with FreeSwitch and a Digium TE220 card (which is
connected to a 4ess T1). Every thing works fine so far except one little
problem that annoys me. Every time I dial in the ivrs, there is a very short
(0.5 second) ringback tone exactly while the "<action
application="answer"/>" line is executed in the dial plan. I want to disable
this ringback tone because the ivrs will answer the call immediately, the
ringback tone is really unnecessary in this case, but I fail to do that. I
try to set the variable "ringback" to change the ringback tone, it's also
not working. Could anyone please help.

I'm using the native pri stack.
The default.xml containts only below lines, and attached is the log file.

      <extension name="testing">
         <condition field="destination_number" expression="^(6.*)$">
               <action application="sleep" data="3000"/>
               <action application="answer"/>
               <action application="sleep" data="3000"/>
         </condition>
      </extension>

Regards,
Tony

!DSPAM:4be6f37432931620317181!

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

 


_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org <http://www.freeswitch.org/> 




-- 
Anthony Minessale II

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

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

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

_______________________________________________
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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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http://www.freeswitch.org

 

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From jan.berger at video24.no  Tue May 11 17:50:46 2010
From: jan.berger at video24.no (Jan Berger)
Date: Wed, 12 May 2010 02:50:46 +0200
Subject: [Freeswitch-users] short ringback tone when answer
In-Reply-To: <3F7EDFD0AB0C46C7880364C2FA9DA27A@fromage>
References: <j2g191e3b11005090616ndd0b0ef8g37f60c1f1016ab4f@mail.gmail.com><549CFEF87AEDE841A38E9D15EAB4C04C5576CE4DC7@cooper><y2w191e3b11005091937k6ea8d442i798b95555e11e817@mail.gmail.com><AANLkTikFk61Phm9lKYIh33BfsMFPfgFZrO-rivpJzi3v@mail.gmail.com><D0216DB842C84F77999189E6FC9CF38E@fromage><B04A76A2-A198-4E01-95FD-A4BBCF8ED4D5@jerris.com><6265B4F0F9F949759F3154920D7F4416@fromage>
	<1033D8D145DA498C8717210040F0BAC1@dell9400>
	<3F7EDFD0AB0C46C7880364C2FA9DA27A@fromage>
Message-ID: <7EFAA4AA76BB472AB4C1C3222C477AF4@dell9400>

http://www.corp.att.com/cpetesting/pdf/tr41459_99.pdf

 

That is the 4ESS and 5ESS standards

 

Jan

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ron
McLeod
Sent: 12. mai 2010 01:59
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

Jan ?

 

This is not my issue ? I am helping someone in HKG that posted earlier
today, but I can answer some the questions on his behalf.

 

I?m not sure of the make/model of the PSTN switch (he may not know either),
but the interface is T1 and the variant is 4ESS.

 

FreeSWITCH is the terminating side of the call in his application, and is
configured as the Q.931 user.  It does send CALL PROCEEDING message in
response to the SETUP message, but it is delayed because he had a 6 second
sleep as the first action in dial plan and PSTN side timed-out waiting (T303
= 4secs).  The stack should probably send a SETUP ACKNOWLEDGE after it finds
a match in the dial plan, before any actions are executed.

 

The PSTN is the one sending the Signal IE in the SETUP message, not
FreeSWITCH.

 

The stack is FreeSWITCH ISDN (not Sangoma, not libpri).

 

What I would like to understand is -- what does including a Signal IE in the
SETUP message actually mean.  Does it inform the terminating end that the
originating end will play the specified call progression to the calling
party automatically (this appears to be what is happening)?  Is it a request
to the terminating switch of what type of in-band call information it should
send back to the calling party?

 

Ron

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jan
Berger
Sent: Tuesday, May 11, 2010 4:18 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

Ron,

 

What equipment and what Q.931 variant are you connected to? If possible get
a PICS out of them.

 

L3 should respond with SETUPACK on a SETUP to avoid re-send. You can respond
with CallProceeding or Alerting, but SetupAck is designed to be sent back
from L3 to buy L4 time.

 

Also ? you need to disable sending of Signal. It is not a mandatory part of
the stack, but I would like to know what equipment/ISDN variant we are
daling with here.

 

Also ? is this OpenZAP, Libpri or a sangoma stack you use on FreeSWITCH ?

 

Jan

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ron
McLeod
Sent: 11. mai 2010 23:30
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

FreeSWITCH ISDN.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Jerris
Sent: Tuesday, May 11, 2010 11:29 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

which pri stack is this?

 

Mike

 

On May 11, 2010, at 1:35 PM, Ron McLeod wrote:

 

Tony sent me a Q.931 trace off-list and I thought I would post it here any
case anyone has seen this before.

 

The telco side is including a Signal IE in the SETUP message.  I?m not sure
what this means ? that the telco will play alerting pattern 0 back to the
CPE unless otherwise instructed?

 

Other things in the trace:

  * SETUP is sent again because the dial plan was executing sleep() before
anything triggered a CALL PROCEEDING.

  * The switch PRI stack is reflecting back the Signal IE in the ALERTING,
CONNECT, and RELEASE messages and the telco complains that the IE is not
supported.

 

 

 

RX [08 02 32 11 05 a1 04 03 80 90 a2 18 03 a1 83 97 34 01 40 6c 0a 00 81 XX
XX XX XX 30 36 39 34 70 05 80 36 37 30 30]

Message Type: [05] SETUP

IE: [a1] Sending Complete

IE: [04] Bearer Capability

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

IE: [6c] Calling party number - TON unknown/NPI unknown/Presentation
allowed/User-provided, not screened/XXXX0694

IE: [70] Called party number - TON unknown/NPI unknown/6700

 

 

RX [08 02 32 11 05 a1 04 03 80 90 a2 18 03 a1 83 97 34 01 40 6c 0a 00 81 XX
XX XX XX 30 36 39 34 70 05 80 36 37 30 30]

Message Type: [05] SETUP

IE: [a1] Sending Complete

IE: [04] Bearer Capability

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

IE: [6c] Calling party number - TON unknown/NPI unknown/Presentation
allowed/User-provided, not screened/XXXX0694

IE: [70] Called party number - TON unknown/NPI unknown/6700

 

 

TX [08 02 b2 11 02 04 03 80 90 a2 18 03 a1 83 97]

Message Type: [02] CALL PROCEEDING

IE: [04] Bearer Capability

IE: [18] Channel identification

 

 

TX [08 02 b2 11 01 04 03 80 90 a2 18 03 a1 83 97 34 01 40]

Message Type: [01] ALERTING

IE: [04] Bearer Capability

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

 

 

TX [08 02 b2 11 07 18 03 a1 83 97 34 01 40]

Message Type: [07] CONNECT

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

 

 

RX [08 02 32 11 7d 08 07 82 e3 34 00 00 00 00 14 01 07]

Message Type: [7d] STATUS

IE: [08] Cause -- IE not implemented/Signal

IE: [14] Call State -- call received

 

 

RX [08 02 32 11 7d 08 07 82 e3 34 00 00 00 00 14 01 08]

Message Type: [7d] STATUS

IE: [08] Cause -- IE not implemented/Signal

IE: [14] Call State -- connect request

 

 

RX [08 02 32 11 0f]

Message Type: [0f] CONNECT ACKNOWLEDGE

 

 

RX [08 02 32 11 45 08 02 82 90]

Message Type: [45] DISCONNECT

IE: [08] Cause -- Normal call clearing

 

 

TX [08 02 b2 11 4d 34 01 40]

Message Type: [4d] RELEASE

IE: [34] Signal - Alerting on ? pattern 0

 

 

RX [08 02 32 11 5a 08 07 82 e3 34 00 00 00 00]

Message Type: [5a] RELEASE COMPLETE

IE: [08] Cause -- IE not implemented/Signal

 

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, May 10, 2010 9:15 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

if ringback is not set, it would never play a tone and even if it was, it
would not be played if you explicitly called answer.

Try reversing the answer and the sleep in your dialplan.

 

 

 

On Sun, May 9, 2010 at 9:37 PM, Tony Tin <tony.tin at noahmedia.com.hk> wrote:

Hi Peter,

Thanks for your reply.

The other end is the Telco, I've confirmed with them that they don't play
the tone for us, also I got the Asterisk install in the same box, it doesn't
play the tone. The tone is played exactly when the answer function is
executed, if I put sleep 10 seconds before the answer, the tone will delay
10 seconds too, any idea?

Regards,
Tony

 

On Mon, May 10, 2010 at 2:33 AM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:

The equipment connected in the other en of the T1 is probably playing these
tones for you, 500ms is probably the time it takes for the called to be
connected successfully.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org
[freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Tony Tin
[tony.tin at noahmedia.com.hk]
Skickat: den 9 maj 2010 15:16
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] short ringback tone when answer


Hi All,

I've just setup a new IVRS with FreeSwitch and a Digium TE220 card (which is
connected to a 4ess T1). Every thing works fine so far except one little
problem that annoys me. Every time I dial in the ivrs, there is a very short
(0.5 second) ringback tone exactly while the "<action
application="answer"/>" line is executed in the dial plan. I want to disable
this ringback tone because the ivrs will answer the call immediately, the
ringback tone is really unnecessary in this case, but I fail to do that. I
try to set the variable "ringback" to change the ringback tone, it's also
not working. Could anyone please help.

I'm using the native pri stack.
The default.xml containts only below lines, and attached is the log file.

      <extension name="testing">
         <condition field="destination_number" expression="^(6.*)$">
               <action application="sleep" data="3000"/>
               <action application="answer"/>
               <action application="sleep" data="3000"/>
         </condition>
      </extension>

Regards,
Tony

!DSPAM:4be6f37432931620317181!

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

 


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




-- 
Anthony Minessale II

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

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

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

_______________________________________________
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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From jan.berger at video24.no  Tue May 11 18:01:11 2010
From: jan.berger at video24.no (Jan Berger)
Date: Wed, 12 May 2010 03:01:11 +0200
Subject: [Freeswitch-users] short ringback tone when answer
In-Reply-To: <867B555E693B4B71AC3D81EDD57DC750@fromage>
References: <j2g191e3b11005090616ndd0b0ef8g37f60c1f1016ab4f@mail.gmail.com><549CFEF87AEDE841A38E9D15EAB4C04C5576CE4DC7@cooper><y2w191e3b11005091937k6ea8d442i798b95555e11e817@mail.gmail.com><AANLkTikFk61Phm9lKYIh33BfsMFPfgFZrO-rivpJzi3v@mail.gmail.com><D0216DB842C84F77999189E6FC9CF38E@fromage><B04A76A2-A198-4E01-95FD-A4BBCF8ED4D5@jerris.com><6265B4F0F9F949759F3154920D7F4416@fromage><1033D8D145DA498C8717210040F0BAC1@dell9400><3F7EDFD0AB0C46C7880364C2FA9DA27A@fromage>
	<F2F9159C1CD04D4DB1E72114507EDCE7@dell9400>
	<867B555E693B4B71AC3D81EDD57DC750@fromage>
Message-ID: <E12A6B7467204C19AA38DDC10B9167A9@dell9400>

Well, the doc?s i sent you don?t even contain the Signal IE 


 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ron
McLeod
Sent: 12. mai 2010 02:48
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

I have the docs ? they are pretty much the same as the ITU docs except for
some additions for propagating traveling class marks and other proprietary
stuff.  I don?t think this issue is variant-specific anyway.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jan
Berger
Sent: Tuesday, May 11, 2010 5:29 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

4ESS means the make is AT&T ? it?s an old legacy thing ? ISDN PRI only. You
can get the standard doc from AT&T for free ? at least it used to be on
their site.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ron
McLeod
Sent: 12. mai 2010 01:59
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

Jan ?

 

This is not my issue ? I am helping someone in HKG that posted earlier
today, but I can answer some the questions on his behalf.

 

I?m not sure of the make/model of the PSTN switch (he may not know either),
but the interface is T1 and the variant is 4ESS.

 

FreeSWITCH is the terminating side of the call in his application, and is
configured as the Q.931 user.  It does send CALL PROCEEDING message in
response to the SETUP message, but it is delayed because he had a 6 second
sleep as the first action in dial plan and PSTN side timed-out waiting (T303
= 4secs).  The stack should probably send a SETUP ACKNOWLEDGE after it finds
a match in the dial plan, before any actions are executed.

 

The PSTN is the one sending the Signal IE in the SETUP message, not
FreeSWITCH.

 

The stack is FreeSWITCH ISDN (not Sangoma, not libpri).

 

What I would like to understand is -- what does including a Signal IE in the
SETUP message actually mean.  Does it inform the terminating end that the
originating end will play the specified call progression to the calling
party automatically (this appears to be what is happening)?  Is it a request
to the terminating switch of what type of in-band call information it should
send back to the calling party?

 

Ron

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jan
Berger
Sent: Tuesday, May 11, 2010 4:18 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

Ron,

 

What equipment and what Q.931 variant are you connected to? If possible get
a PICS out of them.

 

L3 should respond with SETUPACK on a SETUP to avoid re-send. You can respond
with CallProceeding or Alerting, but SetupAck is designed to be sent back
from L3 to buy L4 time.

 

Also ? you need to disable sending of Signal. It is not a mandatory part of
the stack, but I would like to know what equipment/ISDN variant we are
daling with here.

 

Also ? is this OpenZAP, Libpri or a sangoma stack you use on FreeSWITCH ?

 

Jan

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ron
McLeod
Sent: 11. mai 2010 23:30
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

FreeSWITCH ISDN.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Jerris
Sent: Tuesday, May 11, 2010 11:29 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

which pri stack is this?

 

Mike

 

On May 11, 2010, at 1:35 PM, Ron McLeod wrote:

 

Tony sent me a Q.931 trace off-list and I thought I would post it here any
case anyone has seen this before.

 

The telco side is including a Signal IE in the SETUP message.  I?m not sure
what this means ? that the telco will play alerting pattern 0 back to the
CPE unless otherwise instructed?

 

Other things in the trace:

  * SETUP is sent again because the dial plan was executing sleep() before
anything triggered a CALL PROCEEDING.

  * The switch PRI stack is reflecting back the Signal IE in the ALERTING,
CONNECT, and RELEASE messages and the telco complains that the IE is not
supported.

 

 

 

RX [08 02 32 11 05 a1 04 03 80 90 a2 18 03 a1 83 97 34 01 40 6c 0a 00 81 XX
XX XX XX 30 36 39 34 70 05 80 36 37 30 30]

Message Type: [05] SETUP

IE: [a1] Sending Complete

IE: [04] Bearer Capability

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

IE: [6c] Calling party number - TON unknown/NPI unknown/Presentation
allowed/User-provided, not screened/XXXX0694

IE: [70] Called party number - TON unknown/NPI unknown/6700

 

 

RX [08 02 32 11 05 a1 04 03 80 90 a2 18 03 a1 83 97 34 01 40 6c 0a 00 81 XX
XX XX XX 30 36 39 34 70 05 80 36 37 30 30]

Message Type: [05] SETUP

IE: [a1] Sending Complete

IE: [04] Bearer Capability

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

IE: [6c] Calling party number - TON unknown/NPI unknown/Presentation
allowed/User-provided, not screened/XXXX0694

IE: [70] Called party number - TON unknown/NPI unknown/6700

 

 

TX [08 02 b2 11 02 04 03 80 90 a2 18 03 a1 83 97]

Message Type: [02] CALL PROCEEDING

IE: [04] Bearer Capability

IE: [18] Channel identification

 

 

TX [08 02 b2 11 01 04 03 80 90 a2 18 03 a1 83 97 34 01 40]

Message Type: [01] ALERTING

IE: [04] Bearer Capability

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

 

 

TX [08 02 b2 11 07 18 03 a1 83 97 34 01 40]

Message Type: [07] CONNECT

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

 

 

RX [08 02 32 11 7d 08 07 82 e3 34 00 00 00 00 14 01 07]

Message Type: [7d] STATUS

IE: [08] Cause -- IE not implemented/Signal

IE: [14] Call State -- call received

 

 

RX [08 02 32 11 7d 08 07 82 e3 34 00 00 00 00 14 01 08]

Message Type: [7d] STATUS

IE: [08] Cause -- IE not implemented/Signal

IE: [14] Call State -- connect request

 

 

RX [08 02 32 11 0f]

Message Type: [0f] CONNECT ACKNOWLEDGE

 

 

RX [08 02 32 11 45 08 02 82 90]

Message Type: [45] DISCONNECT

IE: [08] Cause -- Normal call clearing

 

 

TX [08 02 b2 11 4d 34 01 40]

Message Type: [4d] RELEASE

IE: [34] Signal - Alerting on ? pattern 0

 

 

RX [08 02 32 11 5a 08 07 82 e3 34 00 00 00 00]

Message Type: [5a] RELEASE COMPLETE

IE: [08] Cause -- IE not implemented/Signal

 

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, May 10, 2010 9:15 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

if ringback is not set, it would never play a tone and even if it was, it
would not be played if you explicitly called answer.

Try reversing the answer and the sleep in your dialplan.

 

 

 

On Sun, May 9, 2010 at 9:37 PM, Tony Tin <tony.tin at noahmedia.com.hk> wrote:

Hi Peter,

Thanks for your reply.

The other end is the Telco, I've confirmed with them that they don't play
the tone for us, also I got the Asterisk install in the same box, it doesn't
play the tone. The tone is played exactly when the answer function is
executed, if I put sleep 10 seconds before the answer, the tone will delay
10 seconds too, any idea?

Regards,
Tony

 

On Mon, May 10, 2010 at 2:33 AM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:

The equipment connected in the other en of the T1 is probably playing these
tones for you, 500ms is probably the time it takes for the called to be
connected successfully.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org
[freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Tony Tin
[tony.tin at noahmedia.com.hk]
Skickat: den 9 maj 2010 15:16
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] short ringback tone when answer


Hi All,

I've just setup a new IVRS with FreeSwitch and a Digium TE220 card (which is
connected to a 4ess T1). Every thing works fine so far except one little
problem that annoys me. Every time I dial in the ivrs, there is a very short
(0.5 second) ringback tone exactly while the "<action
application="answer"/>" line is executed in the dial plan. I want to disable
this ringback tone because the ivrs will answer the call immediately, the
ringback tone is really unnecessary in this case, but I fail to do that. I
try to set the variable "ringback" to change the ringback tone, it's also
not working. Could anyone please help.

I'm using the native pri stack.
The default.xml containts only below lines, and attached is the log file.

      <extension name="testing">
         <condition field="destination_number" expression="^(6.*)$">
               <action application="sleep" data="3000"/>
               <action application="answer"/>
               <action application="sleep" data="3000"/>
         </condition>
      </extension>

Regards,
Tony

!DSPAM:4be6f37432931620317181!

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

 


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




-- 
Anthony Minessale II

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

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

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

_______________________________________________
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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From wchao at yahoo.com  Tue May 11 18:03:05 2010
From: wchao at yahoo.com (Wellie Chao)
Date: Tue, 11 May 2010 21:03:05 -0400 (EDT)
Subject: [Freeswitch-users] Voice to text / speech recognition
In-Reply-To: <AANLkTinEpJxIPc9anUbZEN10UbOxfV0OxSF_ihILi8TC@mail.gmail.com>
References: <alpine.LRH.2.00.1005101227390.20426@osgood.microoffice.com>
	<09FF0B4B64784D0AA874349E55F2C0C9@dell9400>
	<AANLkTinEpJxIPc9anUbZEN10UbOxfV0OxSF_ihILi8TC@mail.gmail.com>
Message-ID: <alpine.LRH.2.00.1005112100330.20426@osgood.microoffice.com>

Thanks! I am installing Vestec's solution now. It only handles small 
vocabulary speech recognition rather than large vocabulary speech 
recognition, but that's also useful (albeit for a different service). We 
can use it for IVR menus. Still looking for a large vocabulary solution 
similar to Nuance's product. I'd love to hear from anybody on the list who 
has used a large vocabulary speech recognition product.



Date: Mon, 10 May 2010 16:46:12 -0700
From: Michael Collins <msc at freeswitch.org>
Reply-To: freeswitch-users at lists.freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Voice to text / speech recognition

You guys need to check out Vestec. Go buy a $25 SDK for your dev server. Tell them that the FS guys sent you and they'll refund
your $25 (check via snail mail). I haven't had a chance to dig into this one yet but I fully intend to do so. However, if you
guys are working on real solutions to real world problems then you're in a great position to give Vestec a test drive. More
info: http://www.freeswitch.org/node/252

See, you should visit freeswitch.org more frequently than twice a year! ;)

-MC

On Mon, May 10, 2010 at 3:49 PM, Jan Berger <jan.berger at video24.no> wrote:
      Hi,

      I am going to look into Sphinx myself for IVR menus.

      Recognition factor is not so much the software as it is the quality of the
      database and how well it is tuned. Tuning you can do yourself with tools
      provided from sphinx + it does exist a separate open source database that I
      believe you can use.

      Doing IVR menus is feasible. But, to translate free speech you have a
      challenge called dialects. The fact is that recognition factor is not 100%
      even with people with the same dialect, and many dialects are very far out
      so analyze the language of your clients and make some very accurate
      questions to the vendor to ensure you get what you want.

      Jan

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Wellie
Chao
Sent: 10. mai 2010 19:09
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Voice to text / speech recognition

I'm looking to provide a feature to transcribe voicemails into text for my
users. I've looked at SpinVox (which was acquired by Nuance). Their
licensing method doesn't seem particularly friendly or flexible. They
charge per user and lock each license to a particular user's actual
identity. If I have a user named John Smith, I have to buy a license from
Nuance _for John Smith_. If John Smith stops using the voicemail to text
feature, but I have another user named Jane Doe who wants to start using
the voicemail to text feature, I have to buy a new license for Jane Doe.
Since Nuance makes you pay for a license one year at a time, it gets
expensive if you have any significant churn. Apart from that, it is just
annoying to have to register users with another company. I'd like to buy
100 or 1000 or 10000 licenses and use them how I see fit and not have to
bother with registering individual users with a provider of speech
recognition services.

I am wondering if other FreeSWITCH users have recommendations for good
voice to text services or software. I'd be happy to consider either a
service or software. If software, ideally it would be free of course, but
a reasonable cost would also be acceptable. I know about PocketSphinx, but
is it really sufficiently high quality as to be useful for transcribing
voicemails? I don't need 100% accuracy -- probably 80% or 90% would be
good enough since I will also attach the WAV file and direct my users to
use the WAV file when in doubt.

Any pointers or tips would be appreciated.

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


_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From ron.freeswitch at mcleodnet.com  Tue May 11 18:18:11 2010
From: ron.freeswitch at mcleodnet.com (Ron McLeod)
Date: Tue, 11 May 2010 18:18:11 -0700
Subject: [Freeswitch-users] short ringback tone when answer
In-Reply-To: <E12A6B7467204C19AA38DDC10B9167A9@dell9400>
References: <j2g191e3b11005090616ndd0b0ef8g37f60c1f1016ab4f@mail.gmail.com><549CFEF87AEDE841A38E9D15EAB4C04C5576CE4DC7@cooper><y2w191e3b11005091937k6ea8d442i798b95555e11e817@mail.gmail.com><AANLkTikFk61Phm9lKYIh33BfsMFPfgFZrO-rivpJzi3v@mail.gmail.com><D0216DB842C84F77999189E6FC9CF38E@fromage><B04A76A2-A198-4E01-95FD-A4BBCF8ED4D5@jerris.com><6265B4F0F9F949759F3154920D7F4416@fromage><1033D8D145DA498C8717210040F0BAC1@dell9400><3F7EDFD0AB0C46C7880364C2FA9DA27A@fromage><F2F9159C1CD04D4DB1E72114507EDCE7@dell9400><867B555E693B4B71AC3D81EDD57DC750@fromage>
	<E12A6B7467204C19AA38DDC10B9167A9@dell9400>
Message-ID: <46AF211EECCA444C918BEEDC562A2D5D@fromage>

It?s in here, but doesn?t say too much:

 
http://www.siplabs.net/docs/300%20DPI%20SEARCHABLE/ATT__TR_41449_ISSUE_2__IS
DN_PRI_2-3__300DPI__SEARCHABLE.pdf

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jan
Berger
Sent: Tuesday, May 11, 2010 6:01 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

Well, the doc?s i sent you don?t even contain the Signal IE 


 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ron
McLeod
Sent: 12. mai 2010 02:48
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

I have the docs ? they are pretty much the same as the ITU docs except for
some additions for propagating traveling class marks and other proprietary
stuff.  I don?t think this issue is variant-specific anyway.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jan
Berger
Sent: Tuesday, May 11, 2010 5:29 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

4ESS means the make is AT&T ? it?s an old legacy thing ? ISDN PRI only. You
can get the standard doc from AT&T for free ? at least it used to be on
their site.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ron
McLeod
Sent: 12. mai 2010 01:59
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

Jan ?

 

This is not my issue ? I am helping someone in HKG that posted earlier
today, but I can answer some the questions on his behalf.

 

I?m not sure of the make/model of the PSTN switch (he may not know either),
but the interface is T1 and the variant is 4ESS.

 

FreeSWITCH is the terminating side of the call in his application, and is
configured as the Q.931 user.  It does send CALL PROCEEDING message in
response to the SETUP message, but it is delayed because he had a 6 second
sleep as the first action in dial plan and PSTN side timed-out waiting (T303
= 4secs).  The stack should probably send a SETUP ACKNOWLEDGE after it finds
a match in the dial plan, before any actions are executed.

 

The PSTN is the one sending the Signal IE in the SETUP message, not
FreeSWITCH.

 

The stack is FreeSWITCH ISDN (not Sangoma, not libpri).

 

What I would like to understand is -- what does including a Signal IE in the
SETUP message actually mean.  Does it inform the terminating end that the
originating end will play the specified call progression to the calling
party automatically (this appears to be what is happening)?  Is it a request
to the terminating switch of what type of in-band call information it should
send back to the calling party?

 

Ron

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jan
Berger
Sent: Tuesday, May 11, 2010 4:18 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

Ron,

 

What equipment and what Q.931 variant are you connected to? If possible get
a PICS out of them.

 

L3 should respond with SETUPACK on a SETUP to avoid re-send. You can respond
with CallProceeding or Alerting, but SetupAck is designed to be sent back
from L3 to buy L4 time.

 

Also ? you need to disable sending of Signal. It is not a mandatory part of
the stack, but I would like to know what equipment/ISDN variant we are
daling with here.

 

Also ? is this OpenZAP, Libpri or a sangoma stack you use on FreeSWITCH ?

 

Jan

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ron
McLeod
Sent: 11. mai 2010 23:30
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

FreeSWITCH ISDN.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Jerris
Sent: Tuesday, May 11, 2010 11:29 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

which pri stack is this?

 

Mike

 

On May 11, 2010, at 1:35 PM, Ron McLeod wrote:

 

Tony sent me a Q.931 trace off-list and I thought I would post it here any
case anyone has seen this before.

 

The telco side is including a Signal IE in the SETUP message.  I?m not sure
what this means ? that the telco will play alerting pattern 0 back to the
CPE unless otherwise instructed?

 

Other things in the trace:

  * SETUP is sent again because the dial plan was executing sleep() before
anything triggered a CALL PROCEEDING.

  * The switch PRI stack is reflecting back the Signal IE in the ALERTING,
CONNECT, and RELEASE messages and the telco complains that the IE is not
supported.

 

 

 

RX [08 02 32 11 05 a1 04 03 80 90 a2 18 03 a1 83 97 34 01 40 6c 0a 00 81 XX
XX XX XX 30 36 39 34 70 05 80 36 37 30 30]

Message Type: [05] SETUP

IE: [a1] Sending Complete

IE: [04] Bearer Capability

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

IE: [6c] Calling party number - TON unknown/NPI unknown/Presentation
allowed/User-provided, not screened/XXXX0694

IE: [70] Called party number - TON unknown/NPI unknown/6700

 

 

RX [08 02 32 11 05 a1 04 03 80 90 a2 18 03 a1 83 97 34 01 40 6c 0a 00 81 XX
XX XX XX 30 36 39 34 70 05 80 36 37 30 30]

Message Type: [05] SETUP

IE: [a1] Sending Complete

IE: [04] Bearer Capability

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

IE: [6c] Calling party number - TON unknown/NPI unknown/Presentation
allowed/User-provided, not screened/XXXX0694

IE: [70] Called party number - TON unknown/NPI unknown/6700

 

 

TX [08 02 b2 11 02 04 03 80 90 a2 18 03 a1 83 97]

Message Type: [02] CALL PROCEEDING

IE: [04] Bearer Capability

IE: [18] Channel identification

 

 

TX [08 02 b2 11 01 04 03 80 90 a2 18 03 a1 83 97 34 01 40]

Message Type: [01] ALERTING

IE: [04] Bearer Capability

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

 

 

TX [08 02 b2 11 07 18 03 a1 83 97 34 01 40]

Message Type: [07] CONNECT

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

 

 

RX [08 02 32 11 7d 08 07 82 e3 34 00 00 00 00 14 01 07]

Message Type: [7d] STATUS

IE: [08] Cause -- IE not implemented/Signal

IE: [14] Call State -- call received

 

 

RX [08 02 32 11 7d 08 07 82 e3 34 00 00 00 00 14 01 08]

Message Type: [7d] STATUS

IE: [08] Cause -- IE not implemented/Signal

IE: [14] Call State -- connect request

 

 

RX [08 02 32 11 0f]

Message Type: [0f] CONNECT ACKNOWLEDGE

 

 

RX [08 02 32 11 45 08 02 82 90]

Message Type: [45] DISCONNECT

IE: [08] Cause -- Normal call clearing

 

 

TX [08 02 b2 11 4d 34 01 40]

Message Type: [4d] RELEASE

IE: [34] Signal - Alerting on ? pattern 0

 

 

RX [08 02 32 11 5a 08 07 82 e3 34 00 00 00 00]

Message Type: [5a] RELEASE COMPLETE

IE: [08] Cause -- IE not implemented/Signal

 

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, May 10, 2010 9:15 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

if ringback is not set, it would never play a tone and even if it was, it
would not be played if you explicitly called answer.

Try reversing the answer and the sleep in your dialplan.

 

 

 

On Sun, May 9, 2010 at 9:37 PM, Tony Tin <tony.tin at noahmedia.com.hk> wrote:

Hi Peter,

Thanks for your reply.

The other end is the Telco, I've confirmed with them that they don't play
the tone for us, also I got the Asterisk install in the same box, it doesn't
play the tone. The tone is played exactly when the answer function is
executed, if I put sleep 10 seconds before the answer, the tone will delay
10 seconds too, any idea?

Regards,
Tony

 

On Mon, May 10, 2010 at 2:33 AM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:

The equipment connected in the other en of the T1 is probably playing these
tones for you, 500ms is probably the time it takes for the called to be
connected successfully.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org
[freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Tony Tin
[tony.tin at noahmedia.com.hk]
Skickat: den 9 maj 2010 15:16
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] short ringback tone when answer


Hi All,

I've just setup a new IVRS with FreeSwitch and a Digium TE220 card (which is
connected to a 4ess T1). Every thing works fine so far except one little
problem that annoys me. Every time I dial in the ivrs, there is a very short
(0.5 second) ringback tone exactly while the "<action
application="answer"/>" line is executed in the dial plan. I want to disable
this ringback tone because the ivrs will answer the call immediately, the
ringback tone is really unnecessary in this case, but I fail to do that. I
try to set the variable "ringback" to change the ringback tone, it's also
not working. Could anyone please help.

I'm using the native pri stack.
The default.xml containts only below lines, and attached is the log file.

      <extension name="testing">
         <condition field="destination_number" expression="^(6.*)$">
               <action application="sleep" data="3000"/>
               <action application="answer"/>
               <action application="sleep" data="3000"/>
         </condition>
      </extension>

Regards,
Tony

!DSPAM:4be6f37432931620317181!

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

 


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




-- 
Anthony Minessale II

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

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

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

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

 

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From mike at jerris.com  Tue May 11 18:19:03 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 11 May 2010 21:19:03 -0400
Subject: [Freeswitch-users] short ringback tone when answer
In-Reply-To: <E12A6B7467204C19AA38DDC10B9167A9@dell9400>
References: <j2g191e3b11005090616ndd0b0ef8g37f60c1f1016ab4f@mail.gmail.com><549CFEF87AEDE841A38E9D15EAB4C04C5576CE4DC7@cooper><y2w191e3b11005091937k6ea8d442i798b95555e11e817@mail.gmail.com><AANLkTikFk61Phm9lKYIh33BfsMFPfgFZrO-rivpJzi3v@mail.gmail.com><D0216DB842C84F77999189E6FC9CF38E@fromage><B04A76A2-A198-4E01-95FD-A4BBCF8ED4D5@jerris.com><6265B4F0F9F949759F3154920D7F4416@fromage><1033D8D145DA498C8717210040F0BAC1@dell9400><3F7EDFD0AB0C46C7880364C2FA9DA27A@fromage>
	<F2F9159C1CD04D4DB1E72114507EDCE7@dell9400>
	<867B555E693B4B71AC3D81EDD57DC750@fromage>
	<E12A6B7467204C19AA38DDC10B9167A9@dell9400>
Message-ID: <65EA73C7-5B40-493E-9AA8-41508F92CFEE@jerris.com>

We can fix this easily enough by never sending out that IE in this dialect.  

Mike

On May 11, 2010, at 9:01 PM, Jan Berger wrote:

> Well, the doc?s i sent you don?t even contain the Signal IE ?
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ron McLeod
> Sent: 12. mai 2010 02:48
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] short ringback tone when answer
>  
> I have the docs ? they are pretty much the same as the ITU docs except for some additions for propagating traveling class marks and other proprietary stuff.  I don?t think this issue is variant-specific anyway.
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jan Berger
> Sent: Tuesday, May 11, 2010 5:29 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] short ringback tone when answer
>  
> 4ESS means the make is AT&T ? it?s an old legacy thing ? ISDN PRI only. You can get the standard doc from AT&T for free ? at least it used to be on their site.
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ron McLeod
> Sent: 12. mai 2010 01:59
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] short ringback tone when answer
>  
> Jan ?
>  
> This is not my issue ? I am helping someone in HKG that posted earlier today, but I can answer some the questions on his behalf.
>  
> I?m not sure of the make/model of the PSTN switch (he may not know either), but the interface is T1 and the variant is 4ESS.
>  
> FreeSWITCH is the terminating side of the call in his application, and is configured as the Q.931 user.  It does send CALL PROCEEDING message in response to the SETUP message, but it is delayed because he had a 6 second sleep as the first action in dial plan and PSTN side timed-out waiting (T303 = 4secs).  The stack should probably send a SETUP ACKNOWLEDGE after it finds a match in the dial plan, before any actions are executed.
>  
> The PSTN is the one sending the Signal IE in the SETUP message, not FreeSWITCH.
>  
> The stack is FreeSWITCH ISDN (not Sangoma, not libpri).
>  
> What I would like to understand is -- what does including a Signal IE in the SETUP message actually mean.  Does it inform the terminating end that the originating end will play the specified call progression to the calling party automatically (this appears to be what is happening)?  Is it a request to the terminating switch of what type of in-band call information it should send back to the calling party?
>  
> Ron
>  
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jan Berger
> Sent: Tuesday, May 11, 2010 4:18 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] short ringback tone when answer
>  
> Ron,
>  
> What equipment and what Q.931 variant are you connected to? If possible get a PICS out of them.
>  
> L3 should respond with SETUPACK on a SETUP to avoid re-send. You can respond with CallProceeding or Alerting, but SetupAck is designed to be sent back from L3 to buy L4 time.
>  
> Also ? you need to disable sending of Signal. It is not a mandatory part of the stack, but I would like to know what equipment/ISDN variant we are daling with here.
>  
> Also ? is this OpenZAP, Libpri or a sangoma stack you use on FreeSWITCH ?
>  
> Jan
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ron McLeod
> Sent: 11. mai 2010 23:30
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] short ringback tone when answer
>  
> FreeSWITCH ISDN.
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
> Sent: Tuesday, May 11, 2010 11:29 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] short ringback tone when answer
>  
> which pri stack is this?
>  
> Mike
>  
> On May 11, 2010, at 1:35 PM, Ron McLeod wrote:
>  
> 
> Tony sent me a Q.931 trace off-list and I thought I would post it here any case anyone has seen this before.
>  
> The telco side is including a Signal IE in the SETUP message.  I?m not sure what this means ? that the telco will play alerting pattern 0 back to the CPE unless otherwise instructed?
>  
> Other things in the trace:
>   * SETUP is sent again because the dial plan was executing sleep() before anything triggered a CALL PROCEEDING.
>   * The switch PRI stack is reflecting back the Signal IE in the ALERTING, CONNECT, and RELEASE messages and the telco complains that the IE is not supported.
>  
>  
>  
> RX [08 02 32 11 05 a1 04 03 80 90 a2 18 03 a1 83 97 34 01 40 6c 0a 00 81 XX XX XX XX 30 36 39 34 70 05 80 36 37 30 30]
> Message Type: [05] SETUP
> IE: [a1] Sending Complete
> IE: [04] Bearer Capability
> IE: [18] Channel identification
> IE: [34] Signal - Alerting on ? pattern 0
> IE: [6c] Calling party number - TON unknown/NPI unknown/Presentation allowed/User-provided, not screened/XXXX0694
> IE: [70] Called party number - TON unknown/NPI unknown/6700
>  
>  
> RX [08 02 32 11 05 a1 04 03 80 90 a2 18 03 a1 83 97 34 01 40 6c 0a 00 81 XX XX XX XX 30 36 39 34 70 05 80 36 37 30 30]
> Message Type: [05] SETUP
> IE: [a1] Sending Complete
> IE: [04] Bearer Capability
> IE: [18] Channel identification
> IE: [34] Signal - Alerting on ? pattern 0
> IE: [6c] Calling party number - TON unknown/NPI unknown/Presentation allowed/User-provided, not screened/XXXX0694
> IE: [70] Called party number - TON unknown/NPI unknown/6700
>  
>  
> TX [08 02 b2 11 02 04 03 80 90 a2 18 03 a1 83 97]
> Message Type: [02] CALL PROCEEDING
> IE: [04] Bearer Capability
> IE: [18] Channel identification
>  
>  
> TX [08 02 b2 11 01 04 03 80 90 a2 18 03 a1 83 97 34 01 40]
> Message Type: [01] ALERTING
> IE: [04] Bearer Capability
> IE: [18] Channel identification
> IE: [34] Signal - Alerting on ? pattern 0
>  
>  
> TX [08 02 b2 11 07 18 03 a1 83 97 34 01 40]
> Message Type: [07] CONNECT
> IE: [18] Channel identification
> IE: [34] Signal - Alerting on ? pattern 0
>  
>  
> RX [08 02 32 11 7d 08 07 82 e3 34 00 00 00 00 14 01 07]
> Message Type: [7d] STATUS
> IE: [08] Cause -- IE not implemented/Signal
> IE: [14] Call State -- call received
>  
>  
> RX [08 02 32 11 7d 08 07 82 e3 34 00 00 00 00 14 01 08]
> Message Type: [7d] STATUS
> IE: [08] Cause -- IE not implemented/Signal
> IE: [14] Call State -- connect request
>  
>  
> RX [08 02 32 11 0f]
> Message Type: [0f] CONNECT ACKNOWLEDGE
>  
>  
> RX [08 02 32 11 45 08 02 82 90]
> Message Type: [45] DISCONNECT
> IE: [08] Cause -- Normal call clearing
>  
>  
> TX [08 02 b2 11 4d 34 01 40]
> Message Type: [4d] RELEASE
> IE: [34] Signal - Alerting on ? pattern 0
>  
>  
> RX [08 02 32 11 5a 08 07 82 e3 34 00 00 00 00]
> Message Type: [5a] RELEASE COMPLETE
> IE: [08] Cause -- IE not implemented/Signal
>  
>  
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale
> Sent: Monday, May 10, 2010 9:15 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] short ringback tone when answer
>  
> if ringback is not set, it would never play a tone and even if it was, it would not be played if you explicitly called answer.
> Try reversing the answer and the sleep in your dialplan.
>  
>  
>  
> 
> On Sun, May 9, 2010 at 9:37 PM, Tony Tin <tony.tin at noahmedia.com.hk> wrote:
> Hi Peter,
> 
> Thanks for your reply.
> 
> The other end is the Telco, I've confirmed with them that they don't play the tone for us, also I got the Asterisk install in the same box, it doesn't play the tone. The tone is played exactly when the answer function is executed, if I put sleep 10 seconds before the answer, the tone will delay 10 seconds too, any idea?
> 
> Regards,
> Tony
>  
> 
> On Mon, May 10, 2010 at 2:33 AM, Peter Olsson <peter.olsson at visionutveckling.se> wrote:
> The equipment connected in the other en of the T1 is probably playing these tones for you, 500ms is probably the time it takes for the called to be connected successfully.
> 
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Tony Tin [tony.tin at noahmedia.com.hk]
> Skickat: den 9 maj 2010 15:16
> Till: freeswitch-users at lists.freeswitch.org
> ?mne: [Freeswitch-users] short ringback tone when answer
> 
> Hi All,
> 
> I've just setup a new IVRS with FreeSwitch and a Digium TE220 card (which is connected to a 4ess T1). Every thing works fine so far except one little problem that annoys me. Every time I dial in the ivrs, there is a very short (0.5 second) ringback tone exactly while the "<action application="answer"/>" line is executed in the dial plan. I want to disable this ringback tone because the ivrs will answer the call immediately, the ringback tone is really unnecessary in this case, but I fail to do that. I try to set the variable "ringback" to change the ringback tone, it's also not working. Could anyone please help.
> 
> I'm using the native pri stack.
> The default.xml containts only below lines, and attached is the log file.
> 
>       <extension name="testing">
>          <condition field="destination_number" expression="^(6.*)$">
>                <action application="sleep" data="3000"/>
>                <action application="answer"/>
>                <action application="sleep" data="3000"/>
>          </condition>
>       </extension>
> 
> Regards,
> Tony
> 
> !DSPAM:4be6f37432931620317181!
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>  
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jan.berger at video24.no  Tue May 11 18:30:17 2010
From: jan.berger at video24.no (Jan Berger)
Date: Wed, 12 May 2010 03:30:17 +0200
Subject: [Freeswitch-users] short ringback tone when answer
In-Reply-To: <867B555E693B4B71AC3D81EDD57DC750@fromage>
References: <j2g191e3b11005090616ndd0b0ef8g37f60c1f1016ab4f@mail.gmail.com><549CFEF87AEDE841A38E9D15EAB4C04C5576CE4DC7@cooper><y2w191e3b11005091937k6ea8d442i798b95555e11e817@mail.gmail.com><AANLkTikFk61Phm9lKYIh33BfsMFPfgFZrO-rivpJzi3v@mail.gmail.com><D0216DB842C84F77999189E6FC9CF38E@fromage><B04A76A2-A198-4E01-95FD-A4BBCF8ED4D5@jerris.com><6265B4F0F9F949759F3154920D7F4416@fromage><1033D8D145DA498C8717210040F0BAC1@dell9400><3F7EDFD0AB0C46C7880364C2FA9DA27A@fromage>
	<F2F9159C1CD04D4DB1E72114507EDCE7@dell9400>
	<867B555E693B4B71AC3D81EDD57DC750@fromage>
Message-ID: <544DE44B7C9C4985BDDDFE95F85C4C31@dell9400>

Just checked the code ? it should be a autoSetupAck flag in the trunk config
that causes the SetupAck to be sent.

 

But, I also see that 5ESS don?t include SIGNAL ? but the stack is changed to
use a generic message where pointer to SIGNAL is included, and not sure if
this struct get zeroed out. 

 

But, you should in my opinion not get the SIGNAL IE in the first place ?
however ? you need tp know the spec reference of the 4ESS as a full
reference is spec name + date ? the specs change over time as well.

 

Jan

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ron
McLeod
Sent: 12. mai 2010 02:48
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

I have the docs ? they are pretty much the same as the ITU docs except for
some additions for propagating traveling class marks and other proprietary
stuff.  I don?t think this issue is variant-specific anyway.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jan
Berger
Sent: Tuesday, May 11, 2010 5:29 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

4ESS means the make is AT&T ? it?s an old legacy thing ? ISDN PRI only. You
can get the standard doc from AT&T for free ? at least it used to be on
their site.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ron
McLeod
Sent: 12. mai 2010 01:59
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

Jan ?

 

This is not my issue ? I am helping someone in HKG that posted earlier
today, but I can answer some the questions on his behalf.

 

I?m not sure of the make/model of the PSTN switch (he may not know either),
but the interface is T1 and the variant is 4ESS.

 

FreeSWITCH is the terminating side of the call in his application, and is
configured as the Q.931 user.  It does send CALL PROCEEDING message in
response to the SETUP message, but it is delayed because he had a 6 second
sleep as the first action in dial plan and PSTN side timed-out waiting (T303
= 4secs).  The stack should probably send a SETUP ACKNOWLEDGE after it finds
a match in the dial plan, before any actions are executed.

 

The PSTN is the one sending the Signal IE in the SETUP message, not
FreeSWITCH.

 

The stack is FreeSWITCH ISDN (not Sangoma, not libpri).

 

What I would like to understand is -- what does including a Signal IE in the
SETUP message actually mean.  Does it inform the terminating end that the
originating end will play the specified call progression to the calling
party automatically (this appears to be what is happening)?  Is it a request
to the terminating switch of what type of in-band call information it should
send back to the calling party?

 

Ron

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jan
Berger
Sent: Tuesday, May 11, 2010 4:18 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

Ron,

 

What equipment and what Q.931 variant are you connected to? If possible get
a PICS out of them.

 

L3 should respond with SETUPACK on a SETUP to avoid re-send. You can respond
with CallProceeding or Alerting, but SetupAck is designed to be sent back
from L3 to buy L4 time.

 

Also ? you need to disable sending of Signal. It is not a mandatory part of
the stack, but I would like to know what equipment/ISDN variant we are
daling with here.

 

Also ? is this OpenZAP, Libpri or a sangoma stack you use on FreeSWITCH ?

 

Jan

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ron
McLeod
Sent: 11. mai 2010 23:30
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

FreeSWITCH ISDN.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Jerris
Sent: Tuesday, May 11, 2010 11:29 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

which pri stack is this?

 

Mike

 

On May 11, 2010, at 1:35 PM, Ron McLeod wrote:

 

Tony sent me a Q.931 trace off-list and I thought I would post it here any
case anyone has seen this before.

 

The telco side is including a Signal IE in the SETUP message.  I?m not sure
what this means ? that the telco will play alerting pattern 0 back to the
CPE unless otherwise instructed?

 

Other things in the trace:

  * SETUP is sent again because the dial plan was executing sleep() before
anything triggered a CALL PROCEEDING.

  * The switch PRI stack is reflecting back the Signal IE in the ALERTING,
CONNECT, and RELEASE messages and the telco complains that the IE is not
supported.

 

 

 

RX [08 02 32 11 05 a1 04 03 80 90 a2 18 03 a1 83 97 34 01 40 6c 0a 00 81 XX
XX XX XX 30 36 39 34 70 05 80 36 37 30 30]

Message Type: [05] SETUP

IE: [a1] Sending Complete

IE: [04] Bearer Capability

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

IE: [6c] Calling party number - TON unknown/NPI unknown/Presentation
allowed/User-provided, not screened/XXXX0694

IE: [70] Called party number - TON unknown/NPI unknown/6700

 

 

RX [08 02 32 11 05 a1 04 03 80 90 a2 18 03 a1 83 97 34 01 40 6c 0a 00 81 XX
XX XX XX 30 36 39 34 70 05 80 36 37 30 30]

Message Type: [05] SETUP

IE: [a1] Sending Complete

IE: [04] Bearer Capability

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

IE: [6c] Calling party number - TON unknown/NPI unknown/Presentation
allowed/User-provided, not screened/XXXX0694

IE: [70] Called party number - TON unknown/NPI unknown/6700

 

 

TX [08 02 b2 11 02 04 03 80 90 a2 18 03 a1 83 97]

Message Type: [02] CALL PROCEEDING

IE: [04] Bearer Capability

IE: [18] Channel identification

 

 

TX [08 02 b2 11 01 04 03 80 90 a2 18 03 a1 83 97 34 01 40]

Message Type: [01] ALERTING

IE: [04] Bearer Capability

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

 

 

TX [08 02 b2 11 07 18 03 a1 83 97 34 01 40]

Message Type: [07] CONNECT

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

 

 

RX [08 02 32 11 7d 08 07 82 e3 34 00 00 00 00 14 01 07]

Message Type: [7d] STATUS

IE: [08] Cause -- IE not implemented/Signal

IE: [14] Call State -- call received

 

 

RX [08 02 32 11 7d 08 07 82 e3 34 00 00 00 00 14 01 08]

Message Type: [7d] STATUS

IE: [08] Cause -- IE not implemented/Signal

IE: [14] Call State -- connect request

 

 

RX [08 02 32 11 0f]

Message Type: [0f] CONNECT ACKNOWLEDGE

 

 

RX [08 02 32 11 45 08 02 82 90]

Message Type: [45] DISCONNECT

IE: [08] Cause -- Normal call clearing

 

 

TX [08 02 b2 11 4d 34 01 40]

Message Type: [4d] RELEASE

IE: [34] Signal - Alerting on ? pattern 0

 

 

RX [08 02 32 11 5a 08 07 82 e3 34 00 00 00 00]

Message Type: [5a] RELEASE COMPLETE

IE: [08] Cause -- IE not implemented/Signal

 

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, May 10, 2010 9:15 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

if ringback is not set, it would never play a tone and even if it was, it
would not be played if you explicitly called answer.

Try reversing the answer and the sleep in your dialplan.

 

 

 

On Sun, May 9, 2010 at 9:37 PM, Tony Tin <tony.tin at noahmedia.com.hk> wrote:

Hi Peter,

Thanks for your reply.

The other end is the Telco, I've confirmed with them that they don't play
the tone for us, also I got the Asterisk install in the same box, it doesn't
play the tone. The tone is played exactly when the answer function is
executed, if I put sleep 10 seconds before the answer, the tone will delay
10 seconds too, any idea?

Regards,
Tony

 

On Mon, May 10, 2010 at 2:33 AM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:

The equipment connected in the other en of the T1 is probably playing these
tones for you, 500ms is probably the time it takes for the called to be
connected successfully.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org
[freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Tony Tin
[tony.tin at noahmedia.com.hk]
Skickat: den 9 maj 2010 15:16
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] short ringback tone when answer


Hi All,

I've just setup a new IVRS with FreeSwitch and a Digium TE220 card (which is
connected to a 4ess T1). Every thing works fine so far except one little
problem that annoys me. Every time I dial in the ivrs, there is a very short
(0.5 second) ringback tone exactly while the "<action
application="answer"/>" line is executed in the dial plan. I want to disable
this ringback tone because the ivrs will answer the call immediately, the
ringback tone is really unnecessary in this case, but I fail to do that. I
try to set the variable "ringback" to change the ringback tone, it's also
not working. Could anyone please help.

I'm using the native pri stack.
The default.xml containts only below lines, and attached is the log file.

      <extension name="testing">
         <condition field="destination_number" expression="^(6.*)$">
               <action application="sleep" data="3000"/>
               <action application="answer"/>
               <action application="sleep" data="3000"/>
         </condition>
      </extension>

Regards,
Tony

!DSPAM:4be6f37432931620317181!

_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org <http://www.freeswitch.org/> 

 


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

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

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

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

_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From rupa at rupa.com  Tue May 11 18:32:18 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 11 May 2010 20:32:18 -0500
Subject: [Freeswitch-users] Mod_lcr question...
In-Reply-To: <AANLkTikL3yFesdq9xkxmYnx2_SD5kTPclUakOHOrj5rE@mail.gmail.com>
References: <AANLkTikL3yFesdq9xkxmYnx2_SD5kTPclUakOHOrj5rE@mail.gmail.com>
Message-ID: <AANLkTikuGUQGGIrdTwYyDP1NczJYhsIMAVfwucEU9ETO@mail.gmail.com>

just ignore that message.  A flag is set on the lcr profile when that table
is not available.  lcr_admin show profiles should tell you more....

On Tue, May 11, 2010 at 7:16 PM, Louis Ntasin <lntasin at gmail.com> wrote:

> Gents-
>
> I am new to this list and would like to say hi to all. I have a question
> about mod_lcr.
>
> When I load mod_lcr I see errors about a missing table:
> npa_nxx_company_ocn. I did some searching and saw that it is meant for intra
> lata/state calling. I don't need this feature. Is there a flag that I can
> turn on to tell mod_lcr not to worry about the table?
>
> Thanks.
>
> Louis
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From ron.freeswitch at mcleodnet.com  Tue May 11 18:33:41 2010
From: ron.freeswitch at mcleodnet.com (Ron McLeod)
Date: Tue, 11 May 2010 18:33:41 -0700
Subject: [Freeswitch-users] short ringback tone when answer
In-Reply-To: <65EA73C7-5B40-493E-9AA8-41508F92CFEE@jerris.com>
References: <j2g191e3b11005090616ndd0b0ef8g37f60c1f1016ab4f@mail.gmail.com><549CFEF87AEDE841A38E9D15EAB4C04C5576CE4DC7@cooper><y2w191e3b11005091937k6ea8d442i798b95555e11e817@mail.gmail.com><AANLkTikFk61Phm9lKYIh33BfsMFPfgFZrO-rivpJzi3v@mail.gmail.com><D0216DB842C84F77999189E6FC9CF38E@fromage><B04A76A2-A198-4E01-95FD-A4BBCF8ED4D5@jerris.com><6265B4F0F9F949759F3154920D7F4416@fromage><1033D8D145DA498C8717210040F0BAC1@dell9400><3F7EDFD0AB0C46C7880364C2FA9DA27A@fromage><F2F9159C1CD04D4DB1E72114507EDCE7@dell9400><867B555E693B4B71AC3D81EDD57DC750@fromage><E12A6B7467204C19AA38DDC10B9167A9@dell9400>
	<65EA73C7-5B40-493E-9AA8-41508F92CFEE@jerris.com>
Message-ID: <7C9AA7A7AF4D47F3B598133E6EECCB76@fromage>

Mike ? that?s not the reported problem.  The issue is that when a call is
delivered from the PSTN to FreeSWITCH, the calling party hears a short of
ringback tone.

 

It would also be good not to send unsupported IE, and as Jan suggests, maybe
send SETUP ACKNOWLEDGEMENT right-away so that the PSTN doesn?t timeout and
re-send the SETUP message.

 

Ron

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Jerris
Sent: Tuesday, May 11, 2010 6:19 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

We can fix this easily enough by never sending out that IE in this dialect.


 

Mike

 

On May 11, 2010, at 9:01 PM, Jan Berger wrote:





Well, the doc?s i sent you don?t even contain the Signal IE 


 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ron
McLeod
Sent: 12. mai 2010 02:48
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

I have the docs ? they are pretty much the same as the ITU docs except for
some additions for propagating traveling class marks and other proprietary
stuff.  I don?t think this issue is variant-specific anyway.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jan
Berger
Sent: Tuesday, May 11, 2010 5:29 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

4ESS means the make is AT&T ? it?s an old legacy thing ? ISDN PRI only. You
can get the standard doc from AT&T for free ? at least it used to be on
their site.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ron
McLeod
Sent: 12. mai 2010 01:59
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

Jan ?

 

This is not my issue ? I am helping someone in HKG that posted earlier
today, but I can answer some the questions on his behalf.

 

I?m not sure of the make/model of the PSTN switch (he may not know either),
but the interface is T1 and the variant is 4ESS.

 

FreeSWITCH is the terminating side of the call in his application, and is
configured as the Q.931 user.  It does send CALL PROCEEDING message in
response to the SETUP message, but it is delayed because he had a 6 second
sleep as the first action in dial plan and PSTN side timed-out waiting (T303
= 4secs).  The stack should probably send a SETUP ACKNOWLEDGE after it finds
a match in the dial plan, before any actions are executed.

 

The PSTN is the one sending the Signal IE in the SETUP message, not
FreeSWITCH.

 

The stack is FreeSWITCH ISDN (not Sangoma, not libpri).

 

What I would like to understand is -- what does including a Signal IE in the
SETUP message actually mean.  Does it inform the terminating end that the
originating end will play the specified call progression to the calling
party automatically (this appears to be what is happening)?  Is it a request
to the terminating switch of what type of in-band call information it should
send back to the calling party?

 

Ron

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jan
Berger
Sent: Tuesday, May 11, 2010 4:18 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

Ron,

 

What equipment and what Q.931 variant are you connected to? If possible get
a PICS out of them.

 

L3 should respond with SETUPACK on a SETUP to avoid re-send. You can respond
with CallProceeding or Alerting, but SetupAck is designed to be sent back
from L3 to buy L4 time.

 

Also ? you need to disable sending of Signal. It is not a mandatory part of
the stack, but I would like to know what equipment/ISDN variant we are
daling with here.

 

Also ? is this OpenZAP, Libpri or a sangoma stack you use on FreeSWITCH ?

 

Jan

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ron
McLeod
Sent: 11. mai 2010 23:30
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

FreeSWITCH ISDN.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Jerris
Sent: Tuesday, May 11, 2010 11:29 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

which pri stack is this?

 

Mike

 

On May 11, 2010, at 1:35 PM, Ron McLeod wrote:

 

Tony sent me a Q.931 trace off-list and I thought I would post it here any
case anyone has seen this before.

 

The telco side is including a Signal IE in the SETUP message.  I?m not sure
what this means ? that the telco will play alerting pattern 0 back to the
CPE unless otherwise instructed?

 

Other things in the trace:

  * SETUP is sent again because the dial plan was executing sleep() before
anything triggered a CALL PROCEEDING.

  * The switch PRI stack is reflecting back the Signal IE in the ALERTING,
CONNECT, and RELEASE messages and the telco complains that the IE is not
supported.

 

 

 

RX [08 02 32 11 05 a1 04 03 80 90 a2 18 03 a1 83 97 34 01 40 6c 0a 00 81 XX
XX XX XX 30 36 39 34 70 05 80 36 37 30 30]

Message Type: [05] SETUP

IE: [a1] Sending Complete

IE: [04] Bearer Capability

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

IE: [6c] Calling party number - TON unknown/NPI unknown/Presentation
allowed/User-provided, not screened/XXXX0694

IE: [70] Called party number - TON unknown/NPI unknown/6700

 

 

RX [08 02 32 11 05 a1 04 03 80 90 a2 18 03 a1 83 97 34 01 40 6c 0a 00 81 XX
XX XX XX 30 36 39 34 70 05 80 36 37 30 30]

Message Type: [05] SETUP

IE: [a1] Sending Complete

IE: [04] Bearer Capability

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

IE: [6c] Calling party number - TON unknown/NPI unknown/Presentation
allowed/User-provided, not screened/XXXX0694

IE: [70] Called party number - TON unknown/NPI unknown/6700

 

 

TX [08 02 b2 11 02 04 03 80 90 a2 18 03 a1 83 97]

Message Type: [02] CALL PROCEEDING

IE: [04] Bearer Capability

IE: [18] Channel identification

 

 

TX [08 02 b2 11 01 04 03 80 90 a2 18 03 a1 83 97 34 01 40]

Message Type: [01] ALERTING

IE: [04] Bearer Capability

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

 

 

TX [08 02 b2 11 07 18 03 a1 83 97 34 01 40]

Message Type: [07] CONNECT

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

 

 

RX [08 02 32 11 7d 08 07 82 e3 34 00 00 00 00 14 01 07]

Message Type: [7d] STATUS

IE: [08] Cause -- IE not implemented/Signal

IE: [14] Call State -- call received

 

 

RX [08 02 32 11 7d 08 07 82 e3 34 00 00 00 00 14 01 08]

Message Type: [7d] STATUS

IE: [08] Cause -- IE not implemented/Signal

IE: [14] Call State -- connect request

 

 

RX [08 02 32 11 0f]

Message Type: [0f] CONNECT ACKNOWLEDGE

 

 

RX [08 02 32 11 45 08 02 82 90]

Message Type: [45] DISCONNECT

IE: [08] Cause -- Normal call clearing

 

 

TX [08 02 b2 11 4d 34 01 40]

Message Type: [4d] RELEASE

IE: [34] Signal - Alerting on ? pattern 0

 

 

RX [08 02 32 11 5a 08 07 82 e3 34 00 00 00 00]

Message Type: [5a] RELEASE COMPLETE

IE: [08] Cause -- IE not implemented/Signal

 

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, May 10, 2010 9:15 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

if ringback is not set, it would never play a tone and even if it was, it
would not be played if you explicitly called answer.

Try reversing the answer and the sleep in your dialplan.

 

 

 

On Sun, May 9, 2010 at 9:37 PM, Tony Tin <tony.tin at noahmedia.com.hk> wrote:

Hi Peter,

Thanks for your reply.

The other end is the Telco, I've confirmed with them that they don't play
the tone for us, also I got the Asterisk install in the same box, it doesn't
play the tone. The tone is played exactly when the answer function is
executed, if I put sleep 10 seconds before the answer, the tone will delay
10 seconds too, any idea?

Regards,
Tony

 

On Mon, May 10, 2010 at 2:33 AM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:

The equipment connected in the other en of the T1 is probably playing these
tones for you, 500ms is probably the time it takes for the called to be
connected successfully.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org
[freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Tony Tin
[tony.tin at noahmedia.com.hk]
Skickat: den 9 maj 2010 15:16
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] short ringback tone when answer


Hi All,

I've just setup a new IVRS with FreeSwitch and a Digium TE220 card (which is
connected to a 4ess T1). Every thing works fine so far except one little
problem that annoys me. Every time I dial in the ivrs, there is a very short
(0.5 second) ringback tone exactly while the "<action
application="answer"/>" line is executed in the dial plan. I want to disable
this ringback tone because the ivrs will answer the call immediately, the
ringback tone is really unnecessary in this case, but I fail to do that. I
try to set the variable "ringback" to change the ringback tone, it's also
not working. Could anyone please help.

I'm using the native pri stack.
The default.xml containts only below lines, and attached is the log file.

      <extension name="testing">
         <condition field="destination_number" expression="^(6.*)$">
               <action application="sleep" data="3000"/>
               <action application="answer"/>
               <action application="sleep" data="3000"/>
         </condition>
      </extension>

Regards,
Tony

!DSPAM:4be6f37432931620317181!

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

 


_______________________________________________
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FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com
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From jan.berger at video24.no  Tue May 11 18:40:38 2010
From: jan.berger at video24.no (Jan Berger)
Date: Wed, 12 May 2010 03:40:38 +0200
Subject: [Freeswitch-users] short ringback tone when answer
In-Reply-To: <65EA73C7-5B40-493E-9AA8-41508F92CFEE@jerris.com>
References: <j2g191e3b11005090616ndd0b0ef8g37f60c1f1016ab4f@mail.gmail.com><549CFEF87AEDE841A38E9D15EAB4C04C5576CE4DC7@cooper><y2w191e3b11005091937k6ea8d442i798b95555e11e817@mail.gmail.com><AANLkTikFk61Phm9lKYIh33BfsMFPfgFZrO-rivpJzi3v@mail.gmail.com><D0216DB842C84F77999189E6FC9CF38E@fromage><B04A76A2-A198-4E01-95FD-A4BBCF8ED4D5@jerris.com><6265B4F0F9F949759F3154920D7F4416@fromage><1033D8D145DA498C8717210040F0BAC1@dell9400><3F7EDFD0AB0C46C7880364C2FA9DA27A@fromage>
	<F2F9159C1CD04D4DB1E72114507EDCE7@dell9400>
	<867B555E693B4B71AC3D81EDD57DC750@fromage>
	<E12A6B7467204C19AA38DDC10B9167A9@dell9400>
	<65EA73C7-5B40-493E-9AA8-41508F92CFEE@jerris.com>
Message-ID: <0F56FCC6711B44349A50F617A5E2CBC5@dell9400>

Yes 

 

And by the way ? the 5ESS need ?some work? :-) - just had a look and closed
the IDE very fast pretending that I did not see what I saw :-)

 

                  case Q931ie_SIGNAL:

                  case Q931ie_CALLING_PARTY_NUMBER:

                  case Q931ie_CALLING_PARTY_SUBADDRESS:

                  case Q931ie_CALLED_PARTY_NUMBER:

                  case Q931ie_CALLED_PARTY_SUBADDRESS:

                  case Q931ie_TRANSIT_NETWORK_SELECTION:

                  case Q931ie_LOW_LAYER_COMPATIBILITY:

                  case Q931ie_HIGH_LAYER_COMPATIBILITY:

                  case Q931ie_FACILITY:

                        rc = Q931Uie[pTrunk->Dialect][IBuf[IOff]](pTrunk,
mes, &IBuf[IOff], &mes->buf[OOff], &IOff, &OOff);

                        if (rc != Q931E_NO_ERROR) 

                             return rc;

                        break;

 

Jan

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Jerris
Sent: 12. mai 2010 03:19
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

We can fix this easily enough by never sending out that IE in this dialect.


 

Mike

 

On May 11, 2010, at 9:01 PM, Jan Berger wrote:





Well, the doc?s i sent you don?t even contain the Signal IE 


 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ron
McLeod
Sent: 12. mai 2010 02:48
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

I have the docs ? they are pretty much the same as the ITU docs except for
some additions for propagating traveling class marks and other proprietary
stuff.  I don?t think this issue is variant-specific anyway.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jan
Berger
Sent: Tuesday, May 11, 2010 5:29 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

4ESS means the make is AT&T ? it?s an old legacy thing ? ISDN PRI only. You
can get the standard doc from AT&T for free ? at least it used to be on
their site.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ron
McLeod
Sent: 12. mai 2010 01:59
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

Jan ?

 

This is not my issue ? I am helping someone in HKG that posted earlier
today, but I can answer some the questions on his behalf.

 

I?m not sure of the make/model of the PSTN switch (he may not know either),
but the interface is T1 and the variant is 4ESS.

 

FreeSWITCH is the terminating side of the call in his application, and is
configured as the Q.931 user.  It does send CALL PROCEEDING message in
response to the SETUP message, but it is delayed because he had a 6 second
sleep as the first action in dial plan and PSTN side timed-out waiting (T303
= 4secs).  The stack should probably send a SETUP ACKNOWLEDGE after it finds
a match in the dial plan, before any actions are executed.

 

The PSTN is the one sending the Signal IE in the SETUP message, not
FreeSWITCH.

 

The stack is FreeSWITCH ISDN (not Sangoma, not libpri).

 

What I would like to understand is -- what does including a Signal IE in the
SETUP message actually mean.  Does it inform the terminating end that the
originating end will play the specified call progression to the calling
party automatically (this appears to be what is happening)?  Is it a request
to the terminating switch of what type of in-band call information it should
send back to the calling party?

 

Ron

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jan
Berger
Sent: Tuesday, May 11, 2010 4:18 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

Ron,

 

What equipment and what Q.931 variant are you connected to? If possible get
a PICS out of them.

 

L3 should respond with SETUPACK on a SETUP to avoid re-send. You can respond
with CallProceeding or Alerting, but SetupAck is designed to be sent back
from L3 to buy L4 time.

 

Also ? you need to disable sending of Signal. It is not a mandatory part of
the stack, but I would like to know what equipment/ISDN variant we are
daling with here.

 

Also ? is this OpenZAP, Libpri or a sangoma stack you use on FreeSWITCH ?

 

Jan

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ron
McLeod
Sent: 11. mai 2010 23:30
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

FreeSWITCH ISDN.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Jerris
Sent: Tuesday, May 11, 2010 11:29 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

which pri stack is this?

 

Mike

 

On May 11, 2010, at 1:35 PM, Ron McLeod wrote:

 

Tony sent me a Q.931 trace off-list and I thought I would post it here any
case anyone has seen this before.

 

The telco side is including a Signal IE in the SETUP message.  I?m not sure
what this means ? that the telco will play alerting pattern 0 back to the
CPE unless otherwise instructed?

 

Other things in the trace:

  * SETUP is sent again because the dial plan was executing sleep() before
anything triggered a CALL PROCEEDING.

  * The switch PRI stack is reflecting back the Signal IE in the ALERTING,
CONNECT, and RELEASE messages and the telco complains that the IE is not
supported.

 

 

 

RX [08 02 32 11 05 a1 04 03 80 90 a2 18 03 a1 83 97 34 01 40 6c 0a 00 81 XX
XX XX XX 30 36 39 34 70 05 80 36 37 30 30]

Message Type: [05] SETUP

IE: [a1] Sending Complete

IE: [04] Bearer Capability

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

IE: [6c] Calling party number - TON unknown/NPI unknown/Presentation
allowed/User-provided, not screened/XXXX0694

IE: [70] Called party number - TON unknown/NPI unknown/6700

 

 

RX [08 02 32 11 05 a1 04 03 80 90 a2 18 03 a1 83 97 34 01 40 6c 0a 00 81 XX
XX XX XX 30 36 39 34 70 05 80 36 37 30 30]

Message Type: [05] SETUP

IE: [a1] Sending Complete

IE: [04] Bearer Capability

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

IE: [6c] Calling party number - TON unknown/NPI unknown/Presentation
allowed/User-provided, not screened/XXXX0694

IE: [70] Called party number - TON unknown/NPI unknown/6700

 

 

TX [08 02 b2 11 02 04 03 80 90 a2 18 03 a1 83 97]

Message Type: [02] CALL PROCEEDING

IE: [04] Bearer Capability

IE: [18] Channel identification

 

 

TX [08 02 b2 11 01 04 03 80 90 a2 18 03 a1 83 97 34 01 40]

Message Type: [01] ALERTING

IE: [04] Bearer Capability

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

 

 

TX [08 02 b2 11 07 18 03 a1 83 97 34 01 40]

Message Type: [07] CONNECT

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

 

 

RX [08 02 32 11 7d 08 07 82 e3 34 00 00 00 00 14 01 07]

Message Type: [7d] STATUS

IE: [08] Cause -- IE not implemented/Signal

IE: [14] Call State -- call received

 

 

RX [08 02 32 11 7d 08 07 82 e3 34 00 00 00 00 14 01 08]

Message Type: [7d] STATUS

IE: [08] Cause -- IE not implemented/Signal

IE: [14] Call State -- connect request

 

 

RX [08 02 32 11 0f]

Message Type: [0f] CONNECT ACKNOWLEDGE

 

 

RX [08 02 32 11 45 08 02 82 90]

Message Type: [45] DISCONNECT

IE: [08] Cause -- Normal call clearing

 

 

TX [08 02 b2 11 4d 34 01 40]

Message Type: [4d] RELEASE

IE: [34] Signal - Alerting on ? pattern 0

 

 

RX [08 02 32 11 5a 08 07 82 e3 34 00 00 00 00]

Message Type: [5a] RELEASE COMPLETE

IE: [08] Cause -- IE not implemented/Signal

 

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, May 10, 2010 9:15 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

if ringback is not set, it would never play a tone and even if it was, it
would not be played if you explicitly called answer.

Try reversing the answer and the sleep in your dialplan.

 

 

 

On Sun, May 9, 2010 at 9:37 PM, Tony Tin <tony.tin at noahmedia.com.hk> wrote:

Hi Peter,

Thanks for your reply.

The other end is the Telco, I've confirmed with them that they don't play
the tone for us, also I got the Asterisk install in the same box, it doesn't
play the tone. The tone is played exactly when the answer function is
executed, if I put sleep 10 seconds before the answer, the tone will delay
10 seconds too, any idea?

Regards,
Tony

 

On Mon, May 10, 2010 at 2:33 AM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:

The equipment connected in the other en of the T1 is probably playing these
tones for you, 500ms is probably the time it takes for the called to be
connected successfully.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org
[freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Tony Tin
[tony.tin at noahmedia.com.hk]
Skickat: den 9 maj 2010 15:16
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] short ringback tone when answer


Hi All,

I've just setup a new IVRS with FreeSwitch and a Digium TE220 card (which is
connected to a 4ess T1). Every thing works fine so far except one little
problem that annoys me. Every time I dial in the ivrs, there is a very short
(0.5 second) ringback tone exactly while the "<action
application="answer"/>" line is executed in the dial plan. I want to disable
this ringback tone because the ivrs will answer the call immediately, the
ringback tone is really unnecessary in this case, but I fail to do that. I
try to set the variable "ringback" to change the ringback tone, it's also
not working. Could anyone please help.

I'm using the native pri stack.
The default.xml containts only below lines, and attached is the log file.

      <extension name="testing">
         <condition field="destination_number" expression="^(6.*)$">
               <action application="sleep" data="3000"/>
               <action application="answer"/>
               <action application="sleep" data="3000"/>
         </condition>
      </extension>

Regards,
Tony

!DSPAM:4be6f37432931620317181!

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

 


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




-- 
Anthony Minessale II

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

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

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<mailto:sip%3A888 at conference.freeswitch.org> 
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pstn:+19193869900

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From jan.berger at video24.no  Tue May 11 18:46:29 2010
From: jan.berger at video24.no (Jan Berger)
Date: Wed, 12 May 2010 03:46:29 +0200
Subject: [Freeswitch-users] short ringback tone when answer
In-Reply-To: <7C9AA7A7AF4D47F3B598133E6EECCB76@fromage>
References: <j2g191e3b11005090616ndd0b0ef8g37f60c1f1016ab4f@mail.gmail.com><549CFEF87AEDE841A38E9D15EAB4C04C5576CE4DC7@cooper><y2w191e3b11005091937k6ea8d442i798b95555e11e817@mail.gmail.com><AANLkTikFk61Phm9lKYIh33BfsMFPfgFZrO-rivpJzi3v@mail.gmail.com><D0216DB842C84F77999189E6FC9CF38E@fromage><B04A76A2-A198-4E01-95FD-A4BBCF8ED4D5@jerris.com><6265B4F0F9F949759F3154920D7F4416@fromage><1033D8D145DA498C8717210040F0BAC1@dell9400><3F7EDFD0AB0C46C7880364C2FA9DA27A@fromage><F2F9159C1CD04D4DB1E72114507EDCE7@dell9400><867B555E693B4B71AC3D81EDD57DC750@fromage><E12A6B7467204C19AA38DDC10B9167A9@dell9400>
	<65EA73C7-5B40-493E-9AA8-41508F92CFEE@jerris.com>
	<7C9AA7A7AF4D47F3B598133E6EECCB76@fromage>
Message-ID: <B9D25A0226504DA7A7D0F11F7D7DB2E5@dell9400>

Is this an in-house switch or a switch in the network?

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ron
McLeod
Sent: 12. mai 2010 03:34
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

Mike ? that?s not the reported problem.  The issue is that when a call is
delivered from the PSTN to FreeSWITCH, the calling party hears a short of
ringback tone.

 

It would also be good not to send unsupported IE, and as Jan suggests, maybe
send SETUP ACKNOWLEDGEMENT right-away so that the PSTN doesn?t timeout and
re-send the SETUP message.

 

Ron

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Jerris
Sent: Tuesday, May 11, 2010 6:19 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

We can fix this easily enough by never sending out that IE in this dialect.


 

Mike

 

On May 11, 2010, at 9:01 PM, Jan Berger wrote:

 

Well, the doc?s i sent you don?t even contain the Signal IE 


 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ron
McLeod
Sent: 12. mai 2010 02:48
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

I have the docs ? they are pretty much the same as the ITU docs except for
some additions for propagating traveling class marks and other proprietary
stuff.  I don?t think this issue is variant-specific anyway.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jan
Berger
Sent: Tuesday, May 11, 2010 5:29 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

4ESS means the make is AT&T ? it?s an old legacy thing ? ISDN PRI only. You
can get the standard doc from AT&T for free ? at least it used to be on
their site.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ron
McLeod
Sent: 12. mai 2010 01:59
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

Jan ?

 

This is not my issue ? I am helping someone in HKG that posted earlier
today, but I can answer some the questions on his behalf.

 

I?m not sure of the make/model of the PSTN switch (he may not know either),
but the interface is T1 and the variant is 4ESS.

 

FreeSWITCH is the terminating side of the call in his application, and is
configured as the Q.931 user.  It does send CALL PROCEEDING message in
response to the SETUP message, but it is delayed because he had a 6 second
sleep as the first action in dial plan and PSTN side timed-out waiting (T303
= 4secs).  The stack should probably send a SETUP ACKNOWLEDGE after it finds
a match in the dial plan, before any actions are executed.

 

The PSTN is the one sending the Signal IE in the SETUP message, not
FreeSWITCH.

 

The stack is FreeSWITCH ISDN (not Sangoma, not libpri).

 

What I would like to understand is -- what does including a Signal IE in the
SETUP message actually mean.  Does it inform the terminating end that the
originating end will play the specified call progression to the calling
party automatically (this appears to be what is happening)?  Is it a request
to the terminating switch of what type of in-band call information it should
send back to the calling party?

 

Ron

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jan
Berger
Sent: Tuesday, May 11, 2010 4:18 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

Ron,

 

What equipment and what Q.931 variant are you connected to? If possible get
a PICS out of them.

 

L3 should respond with SETUPACK on a SETUP to avoid re-send. You can respond
with CallProceeding or Alerting, but SetupAck is designed to be sent back
from L3 to buy L4 time.

 

Also ? you need to disable sending of Signal. It is not a mandatory part of
the stack, but I would like to know what equipment/ISDN variant we are
daling with here.

 

Also ? is this OpenZAP, Libpri or a sangoma stack you use on FreeSWITCH ?

 

Jan

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ron
McLeod
Sent: 11. mai 2010 23:30
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

FreeSWITCH ISDN.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Jerris
Sent: Tuesday, May 11, 2010 11:29 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

which pri stack is this?

 

Mike

 

On May 11, 2010, at 1:35 PM, Ron McLeod wrote:

 

Tony sent me a Q.931 trace off-list and I thought I would post it here any
case anyone has seen this before.

 

The telco side is including a Signal IE in the SETUP message.  I?m not sure
what this means ? that the telco will play alerting pattern 0 back to the
CPE unless otherwise instructed?

 

Other things in the trace:

  * SETUP is sent again because the dial plan was executing sleep() before
anything triggered a CALL PROCEEDING.

  * The switch PRI stack is reflecting back the Signal IE in the ALERTING,
CONNECT, and RELEASE messages and the telco complains that the IE is not
supported.

 

 

 

RX [08 02 32 11 05 a1 04 03 80 90 a2 18 03 a1 83 97 34 01 40 6c 0a 00 81 XX
XX XX XX 30 36 39 34 70 05 80 36 37 30 30]

Message Type: [05] SETUP

IE: [a1] Sending Complete

IE: [04] Bearer Capability

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

IE: [6c] Calling party number - TON unknown/NPI unknown/Presentation
allowed/User-provided, not screened/XXXX0694

IE: [70] Called party number - TON unknown/NPI unknown/6700

 

 

RX [08 02 32 11 05 a1 04 03 80 90 a2 18 03 a1 83 97 34 01 40 6c 0a 00 81 XX
XX XX XX 30 36 39 34 70 05 80 36 37 30 30]

Message Type: [05] SETUP

IE: [a1] Sending Complete

IE: [04] Bearer Capability

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

IE: [6c] Calling party number - TON unknown/NPI unknown/Presentation
allowed/User-provided, not screened/XXXX0694

IE: [70] Called party number - TON unknown/NPI unknown/6700

 

 

TX [08 02 b2 11 02 04 03 80 90 a2 18 03 a1 83 97]

Message Type: [02] CALL PROCEEDING

IE: [04] Bearer Capability

IE: [18] Channel identification

 

 

TX [08 02 b2 11 01 04 03 80 90 a2 18 03 a1 83 97 34 01 40]

Message Type: [01] ALERTING

IE: [04] Bearer Capability

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

 

 

TX [08 02 b2 11 07 18 03 a1 83 97 34 01 40]

Message Type: [07] CONNECT

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

 

 

RX [08 02 32 11 7d 08 07 82 e3 34 00 00 00 00 14 01 07]

Message Type: [7d] STATUS

IE: [08] Cause -- IE not implemented/Signal

IE: [14] Call State -- call received

 

 

RX [08 02 32 11 7d 08 07 82 e3 34 00 00 00 00 14 01 08]

Message Type: [7d] STATUS

IE: [08] Cause -- IE not implemented/Signal

IE: [14] Call State -- connect request

 

 

RX [08 02 32 11 0f]

Message Type: [0f] CONNECT ACKNOWLEDGE

 

 

RX [08 02 32 11 45 08 02 82 90]

Message Type: [45] DISCONNECT

IE: [08] Cause -- Normal call clearing

 

 

TX [08 02 b2 11 4d 34 01 40]

Message Type: [4d] RELEASE

IE: [34] Signal - Alerting on ? pattern 0

 

 

RX [08 02 32 11 5a 08 07 82 e3 34 00 00 00 00]

Message Type: [5a] RELEASE COMPLETE

IE: [08] Cause -- IE not implemented/Signal

 

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, May 10, 2010 9:15 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

if ringback is not set, it would never play a tone and even if it was, it
would not be played if you explicitly called answer.

Try reversing the answer and the sleep in your dialplan.

 

 

 

On Sun, May 9, 2010 at 9:37 PM, Tony Tin <tony.tin at noahmedia.com.hk> wrote:

Hi Peter,

Thanks for your reply.

The other end is the Telco, I've confirmed with them that they don't play
the tone for us, also I got the Asterisk install in the same box, it doesn't
play the tone. The tone is played exactly when the answer function is
executed, if I put sleep 10 seconds before the answer, the tone will delay
10 seconds too, any idea?

Regards,
Tony

 

On Mon, May 10, 2010 at 2:33 AM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:

The equipment connected in the other en of the T1 is probably playing these
tones for you, 500ms is probably the time it takes for the called to be
connected successfully.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org
[freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Tony Tin
[tony.tin at noahmedia.com.hk]
Skickat: den 9 maj 2010 15:16
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] short ringback tone when answer


Hi All,

I've just setup a new IVRS with FreeSwitch and a Digium TE220 card (which is
connected to a 4ess T1). Every thing works fine so far except one little
problem that annoys me. Every time I dial in the ivrs, there is a very short
(0.5 second) ringback tone exactly while the "<action
application="answer"/>" line is executed in the dial plan. I want to disable
this ringback tone because the ivrs will answer the call immediately, the
ringback tone is really unnecessary in this case, but I fail to do that. I
try to set the variable "ringback" to change the ringback tone, it's also
not working. Could anyone please help.

I'm using the native pri stack.
The default.xml containts only below lines, and attached is the log file.

      <extension name="testing">
         <condition field="destination_number" expression="^(6.*)$">
               <action application="sleep" data="3000"/>
               <action application="answer"/>
               <action application="sleep" data="3000"/>
         </condition>
      </extension>

Regards,
Tony

!DSPAM:4be6f37432931620317181!

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

 


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




-- 
Anthony Minessale II

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MSN:anthony_minessale at hotmail.com
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_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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_______________________________________________
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From info at evestech.com  Tue May 11 18:51:38 2010
From: info at evestech.com (Kashif Kahn)
Date: Tue, 11 May 2010 18:51:38 -0700 (PDT)
Subject: [Freeswitch-users] Voice to text / speech recognition
In-Reply-To: <alpine.LRH.2.00.1005112100330.20426@osgood.microoffice.com>
References: <alpine.LRH.2.00.1005101227390.20426@osgood.microoffice.com>
	<09FF0B4B64784D0AA874349E55F2C0C9@dell9400>
	<AANLkTinEpJxIPc9anUbZEN10UbOxfV0OxSF_ihILi8TC@mail.gmail.com>
	<alpine.LRH.2.00.1005112100330.20426@osgood.microoffice.com>
Message-ID: <559583.695.qm@web204.biz.mail.re2.yahoo.com>

Hi Wellie,

Many thanks for purchasing Vestec speech recognition engine.

Respectfully, I would like to make a few clarifications: 

(a) Vestec speech engine is designed for IVR-type interactive voice applications. Almost all such applications can be easily addressed with a vocabulary size of 500 keywords per recognition. (Remember, there can be several recognitions within one application; each recognition pertains to one state, such as one level in a multi-level menu tree). In addition, a variety of non-IVR applications - such as speech-based directory assistance (ie. utterance of a person's name as opposed to entering the customary first three letters of a person's last name for routing) - can be built for all SMB and most SME applications with a speech engine supporting 500 keywords vocabulary size.

(b) Vestec speech engine is not designed to be used as a general purpose dictation/transcription system. Such dictation/transcription systems - for example, for voicemail-to-email transcription - generally require a vocabulary size of tens of thousands of words, not to mention important "training" data (that must be collected or purchased separately) for building the grammar for the desired application. Nuance offers a dictation system in the form of its its Dragon Speech product.

(c) in the IVR space - which by the way constitutes the overwhelming majority of commercial speech applications and is a natural fit with telephony platforms such as Freeswitch - Vestec speech engine offers the best deal around. Not only does it offer among the highest recognition accuracy in the industry and industry standards-based grammar writing format, it supports the same vocabulary size as LumenVox Lite (500 keywords) but costs less than 50% of its retail price, and offers twice the vocabulary size of Nuance tier-1 engine (250 keywords) and retails at less than 20% of its price.

Hope this helps.

Best regards,
-Kashif

 Kashif Kahn
VP, Business Development
Vestec, Inc.
Waterloo, ON Canada
phone: (519) 885-7615




________________________________
From: Wellie Chao <wchao at yahoo.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Tue, May 11, 2010 9:03:05 PM
Subject: Re: [Freeswitch-users] Voice to text / speech recognition

Thanks! I am installing Vestec's solution now. It only handles small 
vocabulary speech recognition rather than large vocabulary speech 
recognition, but that's also useful (albeit for a different service). We 
can use it for IVR menus. Still looking for a large vocabulary solution 
similar to Nuance's product. I'd love to hear from anybody on the list who 
has used a large vocabulary speech recognition product.



Date: Mon, 10 May 2010 16:46:12 -0700
From: Michael Collins <msc at freeswitch.org>
Reply-To: freeswitch-users at lists.freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Voice to text / speech recognition

You guys need to check out Vestec. Go buy a $25 SDK for your dev server. Tell them that the FS guys sent you and they'll refund
your $25 (check via snail mail). I haven't had a chance to dig into this one yet but I fully intend to do so. However, if you
guys are working on real solutions to real world problems then you're in a great position to give Vestec a test drive. More
info: http://www.freeswitch.org/node/252

See, you should visit freeswitch.org more frequently than twice a year! ;)

-MC

On Mon, May 10, 2010 at 3:49 PM, Jan Berger <jan.berger at video24.no> wrote:
      Hi,

      I am going to look into Sphinx myself for IVR menus.

      Recognition factor is not so much the software as it is the quality of the
      database and how well it is tuned. Tuning you can do yourself with tools
      provided from sphinx + it does exist a separate open source database that I
      believe you can use.

      Doing IVR menus is feasible. But, to translate free speech you have a
      challenge called dialects. The fact is that recognition factor is not 100%
      even with people with the same dialect, and many dialects are very far out
      so analyze the language of your clients and make some very accurate
      questions to the vendor to ensure you get what you want.

      Jan

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Wellie
Chao
Sent: 10. mai 2010 19:09
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Voice to text / speech recognition

I'm looking to provide a feature to transcribe voicemails into text for my
users. I've looked at SpinVox (which was acquired by Nuance). Their
licensing method doesn't seem particularly friendly or flexible. They
charge per user and lock each license to a particular user's actual
identity. If I have a user named John Smith, I have to buy a license from
Nuance _for John Smith_. If John Smith stops using the voicemail to text
feature, but I have another user named Jane Doe who wants to start using
the voicemail to text feature, I have to buy a new license for Jane Doe.
Since Nuance makes you pay for a license one year at a time, it gets
expensive if you have any significant churn. Apart from that, it is just
annoying to have to register users with another company. I'd like to buy
100 or 1000 or 10000 licenses and use them how I see fit and not have to
bother with registering individual users with a provider of speech
recognition services.

I am wondering if other FreeSWITCH users have recommendations for good
voice to text services or software. I'd be happy to consider either a
service or software. If software, ideally it would be free of course, but
a reasonable cost would also be acceptable. I know about PocketSphinx, but
is it really sufficiently high quality as to be useful for transcribing
voicemails? I don't need 100% accuracy -- probably 80% or 90% would be
good enough since I will also attach the WAV file and direct my users to
use the WAV file when in doubt.

Any pointers or tips would be appreciated.

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


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




_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From jan.berger at video24.no  Tue May 11 18:57:31 2010
From: jan.berger at video24.no (Jan Berger)
Date: Wed, 12 May 2010 03:57:31 +0200
Subject: [Freeswitch-users] short ringback tone when answer
In-Reply-To: <46AF211EECCA444C918BEEDC562A2D5D@fromage>
References: <j2g191e3b11005090616ndd0b0ef8g37f60c1f1016ab4f@mail.gmail.com><549CFEF87AEDE841A38E9D15EAB4C04C5576CE4DC7@cooper><y2w191e3b11005091937k6ea8d442i798b95555e11e817@mail.gmail.com><AANLkTikFk61Phm9lKYIh33BfsMFPfgFZrO-rivpJzi3v@mail.gmail.com><D0216DB842C84F77999189E6FC9CF38E@fromage><B04A76A2-A198-4E01-95FD-A4BBCF8ED4D5@jerris.com><6265B4F0F9F949759F3154920D7F4416@fromage><1033D8D145DA498C8717210040F0BAC1@dell9400><3F7EDFD0AB0C46C7880364C2FA9DA27A@fromage><F2F9159C1CD04D4DB1E72114507EDCE7@dell9400><867B555E693B4B71AC3D81EDD57DC750@fromage>
	<E12A6B7467204C19AA38DDC10B9167A9@dell9400>
	<46AF211EECCA444C918BEEDC562A2D5D@fromage>
Message-ID: <5CD35414D3CB4A8BA42F0672E07098E2@dell9400>

Actually it says what it needs ? but the newer version of the spec says n/a
on signal IE, so you are dealing with an older version of the stack.

 

?The purpose information element is to allow the network to optionally
convey information to a user

regarding tones and alerting signals.?

 

Jan

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ron
McLeod
Sent: 12. mai 2010 03:18
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

It?s in here, but doesn?t say too much:

 
http://www.siplabs.net/docs/300%20DPI%20SEARCHABLE/ATT__TR_41449_ISSUE_2__IS
DN_PRI_2-3__300DPI__SEARCHABLE.pdf

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jan
Berger
Sent: Tuesday, May 11, 2010 6:01 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

Well, the doc?s i sent you don?t even contain the Signal IE 


 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ron
McLeod
Sent: 12. mai 2010 02:48
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

I have the docs ? they are pretty much the same as the ITU docs except for
some additions for propagating traveling class marks and other proprietary
stuff.  I don?t think this issue is variant-specific anyway.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jan
Berger
Sent: Tuesday, May 11, 2010 5:29 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

4ESS means the make is AT&T ? it?s an old legacy thing ? ISDN PRI only. You
can get the standard doc from AT&T for free ? at least it used to be on
their site.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ron
McLeod
Sent: 12. mai 2010 01:59
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

Jan ?

 

This is not my issue ? I am helping someone in HKG that posted earlier
today, but I can answer some the questions on his behalf.

 

I?m not sure of the make/model of the PSTN switch (he may not know either),
but the interface is T1 and the variant is 4ESS.

 

FreeSWITCH is the terminating side of the call in his application, and is
configured as the Q.931 user.  It does send CALL PROCEEDING message in
response to the SETUP message, but it is delayed because he had a 6 second
sleep as the first action in dial plan and PSTN side timed-out waiting (T303
= 4secs).  The stack should probably send a SETUP ACKNOWLEDGE after it finds
a match in the dial plan, before any actions are executed.

 

The PSTN is the one sending the Signal IE in the SETUP message, not
FreeSWITCH.

 

The stack is FreeSWITCH ISDN (not Sangoma, not libpri).

 

What I would like to understand is -- what does including a Signal IE in the
SETUP message actually mean.  Does it inform the terminating end that the
originating end will play the specified call progression to the calling
party automatically (this appears to be what is happening)?  Is it a request
to the terminating switch of what type of in-band call information it should
send back to the calling party?

 

Ron

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jan
Berger
Sent: Tuesday, May 11, 2010 4:18 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

Ron,

 

What equipment and what Q.931 variant are you connected to? If possible get
a PICS out of them.

 

L3 should respond with SETUPACK on a SETUP to avoid re-send. You can respond
with CallProceeding or Alerting, but SetupAck is designed to be sent back
from L3 to buy L4 time.

 

Also ? you need to disable sending of Signal. It is not a mandatory part of
the stack, but I would like to know what equipment/ISDN variant we are
daling with here.

 

Also ? is this OpenZAP, Libpri or a sangoma stack you use on FreeSWITCH ?

 

Jan

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ron
McLeod
Sent: 11. mai 2010 23:30
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

FreeSWITCH ISDN.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Jerris
Sent: Tuesday, May 11, 2010 11:29 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

which pri stack is this?

 

Mike

 

On May 11, 2010, at 1:35 PM, Ron McLeod wrote:

 

Tony sent me a Q.931 trace off-list and I thought I would post it here any
case anyone has seen this before.

 

The telco side is including a Signal IE in the SETUP message.  I?m not sure
what this means ? that the telco will play alerting pattern 0 back to the
CPE unless otherwise instructed?

 

Other things in the trace:

  * SETUP is sent again because the dial plan was executing sleep() before
anything triggered a CALL PROCEEDING.

  * The switch PRI stack is reflecting back the Signal IE in the ALERTING,
CONNECT, and RELEASE messages and the telco complains that the IE is not
supported.

 

 

 

RX [08 02 32 11 05 a1 04 03 80 90 a2 18 03 a1 83 97 34 01 40 6c 0a 00 81 XX
XX XX XX 30 36 39 34 70 05 80 36 37 30 30]

Message Type: [05] SETUP

IE: [a1] Sending Complete

IE: [04] Bearer Capability

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

IE: [6c] Calling party number - TON unknown/NPI unknown/Presentation
allowed/User-provided, not screened/XXXX0694

IE: [70] Called party number - TON unknown/NPI unknown/6700

 

 

RX [08 02 32 11 05 a1 04 03 80 90 a2 18 03 a1 83 97 34 01 40 6c 0a 00 81 XX
XX XX XX 30 36 39 34 70 05 80 36 37 30 30]

Message Type: [05] SETUP

IE: [a1] Sending Complete

IE: [04] Bearer Capability

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

IE: [6c] Calling party number - TON unknown/NPI unknown/Presentation
allowed/User-provided, not screened/XXXX0694

IE: [70] Called party number - TON unknown/NPI unknown/6700

 

 

TX [08 02 b2 11 02 04 03 80 90 a2 18 03 a1 83 97]

Message Type: [02] CALL PROCEEDING

IE: [04] Bearer Capability

IE: [18] Channel identification

 

 

TX [08 02 b2 11 01 04 03 80 90 a2 18 03 a1 83 97 34 01 40]

Message Type: [01] ALERTING

IE: [04] Bearer Capability

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

 

 

TX [08 02 b2 11 07 18 03 a1 83 97 34 01 40]

Message Type: [07] CONNECT

IE: [18] Channel identification

IE: [34] Signal - Alerting on ? pattern 0

 

 

RX [08 02 32 11 7d 08 07 82 e3 34 00 00 00 00 14 01 07]

Message Type: [7d] STATUS

IE: [08] Cause -- IE not implemented/Signal

IE: [14] Call State -- call received

 

 

RX [08 02 32 11 7d 08 07 82 e3 34 00 00 00 00 14 01 08]

Message Type: [7d] STATUS

IE: [08] Cause -- IE not implemented/Signal

IE: [14] Call State -- connect request

 

 

RX [08 02 32 11 0f]

Message Type: [0f] CONNECT ACKNOWLEDGE

 

 

RX [08 02 32 11 45 08 02 82 90]

Message Type: [45] DISCONNECT

IE: [08] Cause -- Normal call clearing

 

 

TX [08 02 b2 11 4d 34 01 40]

Message Type: [4d] RELEASE

IE: [34] Signal - Alerting on ? pattern 0

 

 

RX [08 02 32 11 5a 08 07 82 e3 34 00 00 00 00]

Message Type: [5a] RELEASE COMPLETE

IE: [08] Cause -- IE not implemented/Signal

 

 

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Monday, May 10, 2010 9:15 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] short ringback tone when answer

 

if ringback is not set, it would never play a tone and even if it was, it
would not be played if you explicitly called answer.

Try reversing the answer and the sleep in your dialplan.

 

 

 

On Sun, May 9, 2010 at 9:37 PM, Tony Tin <tony.tin at noahmedia.com.hk> wrote:

Hi Peter,

Thanks for your reply.

The other end is the Telco, I've confirmed with them that they don't play
the tone for us, also I got the Asterisk install in the same box, it doesn't
play the tone. The tone is played exactly when the answer function is
executed, if I put sleep 10 seconds before the answer, the tone will delay
10 seconds too, any idea?

Regards,
Tony

 

On Mon, May 10, 2010 at 2:33 AM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:

The equipment connected in the other en of the T1 is probably playing these
tones for you, 500ms is probably the time it takes for the called to be
connected successfully.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org
[freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Tony Tin
[tony.tin at noahmedia.com.hk]
Skickat: den 9 maj 2010 15:16
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] short ringback tone when answer


Hi All,

I've just setup a new IVRS with FreeSwitch and a Digium TE220 card (which is
connected to a 4ess T1). Every thing works fine so far except one little
problem that annoys me. Every time I dial in the ivrs, there is a very short
(0.5 second) ringback tone exactly while the "<action
application="answer"/>" line is executed in the dial plan. I want to disable
this ringback tone because the ivrs will answer the call immediately, the
ringback tone is really unnecessary in this case, but I fail to do that. I
try to set the variable "ringback" to change the ringback tone, it's also
not working. Could anyone please help.

I'm using the native pri stack.
The default.xml containts only below lines, and attached is the log file.

      <extension name="testing">
         <condition field="destination_number" expression="^(6.*)$">
               <action application="sleep" data="3000"/>
               <action application="answer"/>
               <action application="sleep" data="3000"/>
         </condition>
      </extension>

Regards,
Tony

!DSPAM:4be6f37432931620317181!

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

 


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




-- 
Anthony Minessale II

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

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

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

_______________________________________________
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From lntasin at gmail.com  Tue May 11 19:11:23 2010
From: lntasin at gmail.com (Louis Ntasin)
Date: Tue, 11 May 2010 22:11:23 -0400
Subject: [Freeswitch-users] Mod_lcr question...
In-Reply-To: <AANLkTikuGUQGGIrdTwYyDP1NczJYhsIMAVfwucEU9ETO@mail.gmail.com>
References: <AANLkTikL3yFesdq9xkxmYnx2_SD5kTPclUakOHOrj5rE@mail.gmail.com>
	<AANLkTikuGUQGGIrdTwYyDP1NczJYhsIMAVfwucEU9ETO@mail.gmail.com>
Message-ID: <AANLkTim_3qrlGBhqKddpmOSjvMIh76MbSvwrl_b4ftDD@mail.gmail.com>

Thank you.


On Tue, May 11, 2010 at 9:32 PM, Rupa Schomaker <rupa at rupa.com> wrote:

> just ignore that message.  A flag is set on the lcr profile when that table
> is not available.  lcr_admin show profiles should tell you more....
>
> On Tue, May 11, 2010 at 7:16 PM, Louis Ntasin <lntasin at gmail.com> wrote:
>
>> Gents-
>>
>> I am new to this list and would like to say hi to all. I have a question
>> about mod_lcr.
>>
>> When I load mod_lcr I see errors about a missing table:
>> npa_nxx_company_ocn. I did some searching and saw that it is meant for intra
>> lata/state calling. I don't need this feature. Is there a flag that I can
>> turn on to tell mod_lcr not to worry about the table?
>>
>> Thanks.
>>
>> Louis
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From chris.chen2004 at gmail.com  Tue May 11 20:13:13 2010
From: chris.chen2004 at gmail.com (Chris Chen)
Date: Tue, 11 May 2010 23:13:13 -0400
Subject: [Freeswitch-users] Aastra and SCA
In-Reply-To: <1152159774382682281@unknownmsgid>
References: <2A6C7EC2-913D-417E-ADD6-E3A72C6AA6E6@freeswitch.org>
	<28B636AB-D221-44C5-BDA4-1D12AC44748B@freeswitch.org>
	<1152159774382682281@unknownmsgid>
Message-ID: <AANLkTimSSXJkfxHCLkPKYM2XJImyo1-H6dja9Ua7Ajpt@mail.gmail.com>

I just tested the 2.6 firmware on my Aastra 55i, looks like they have made
SCA working with my Polycom phone IP 650 now
I will see if there any other problems with Aastra phone.
Chris Chen


On Tue, May 11, 2010 at 4:12 PM, Mark Maly <mark.maly at molcs.org> wrote:

>  FYI-
>
>
>
> Got a ticket w/Aastra and they said,
>
>
>
> ?We have just released a new GA firmware that I would like for you to load
> onto one of your phones to see if it resolves the issue. You can download
> this from our website at www.aastratelecom.com/support and click on
> Download Area and select the 6731i. The firmware version you are looking for
> is 2.6 and it is listed under Current Software Release.
>
>
>
> Please load that on a test phone and let us know the results.?
>
>
>
> I?ll let you know,
>
> Mark
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West
> *Sent:* Monday, May 03, 2010 8:00 PM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Aastra and SCA
>
>
>
> Nope but we have exchanged some emails about it back and forth and some
> beta firmware where they did half way fix it... but seems you need the
> call-info header on ALL packets associated with the dialog of that call.
>  They are still missing a few.
>
>
>
> /b
>
>
>
> On May 3, 2010, at 7:55 PM, Mark Maly wrote:
>
>
>
>   Brian,
>
>
>
> After my first reply I did receive a msg from Aastra support and whether FS
> had reported the problem.   The contact wondered whether they [FS] had been
> given any ?reference number? related to the problem.
>
>
>
> Would love to help and would pass ref num if you had one.
>
>
>
> Mark
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West
> *Sent:* Sunday, May 02, 2010 11:14 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Aastra and SCA
>
>
>
> Give up their exists a bug that prevents it from working on any aastra.
>  Polycom and Cisco SPA work flawless.
>
>
>
> /b
>
> Sent from my iPad
>
>
> On Apr 30, 2010, at 9:07 PM, "Mark Maly" <mark.maly at molcs.org> wrote:
>
>   Hi,
>
>
>
> I?ve tried to patiently figure this out by reading the wiki and this list.
> Unfortunately, I?ve been unable to get it right.
>
>
>
> I have 2 Aastra 6731is and a 51i and trying to get SCA working.  I?m
> experiencing problems similar to the Cisco thread from last month ? outgoing
> calls implement SCA well.  Incoming calls ring all lines and appearances
> work, but when one phone is answered, the line appearances are removed from
> the remaining phones.  I am not attempting to use any DNS.
>
>
>
> My configuration has all three phones plus FS on a local LAN.  Nothing too
> fancy.  Each line is configured for Broadsoft SCA and SCA bridging  is
> enabled globally for the phones.
>
>
>
> Trying to update/replace an old phone at my church.
>
>
>
> Any help would be greatly appreciated.
>
>
>
> Thanks, Mark
>
> Mark.maly at molcs.org
>
>
>
>   _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>  _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Tue May 11 20:25:51 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 11 May 2010 22:25:51 -0500
Subject: [Freeswitch-users] Aastra and SCA
In-Reply-To: <AANLkTimSSXJkfxHCLkPKYM2XJImyo1-H6dja9Ua7Ajpt@mail.gmail.com>
References: <2A6C7EC2-913D-417E-ADD6-E3A72C6AA6E6@freeswitch.org>
	<28B636AB-D221-44C5-BDA4-1D12AC44748B@freeswitch.org>
	<1152159774382682281@unknownmsgid>
	<AANLkTimSSXJkfxHCLkPKYM2XJImyo1-H6dja9Ua7Ajpt@mail.gmail.com>
Message-ID: <3B09563A-7594-4DD6-A78B-6F7ED4A20A56@freeswitch.org>

make it work with multiple lines now.  ;)

/b

On May 11, 2010, at 10:13 PM, Chris Chen wrote:

> I just tested the 2.6 firmware on my Aastra 55i, looks like they have made SCA working with my Polycom phone IP 650 now
> I will see if there any other problems with Aastra phone.
> Chris Chen
> 



From chris.chen2004 at gmail.com  Tue May 11 20:35:25 2010
From: chris.chen2004 at gmail.com (Chris Chen)
Date: Tue, 11 May 2010 23:35:25 -0400
Subject: [Freeswitch-users] Aastra and SCA
In-Reply-To: <AANLkTimSSXJkfxHCLkPKYM2XJImyo1-H6dja9Ua7Ajpt@mail.gmail.com>
References: <2A6C7EC2-913D-417E-ADD6-E3A72C6AA6E6@freeswitch.org>
	<28B636AB-D221-44C5-BDA4-1D12AC44748B@freeswitch.org>
	<1152159774382682281@unknownmsgid>
	<AANLkTimSSXJkfxHCLkPKYM2XJImyo1-H6dja9Ua7Ajpt@mail.gmail.com>
Message-ID: <AANLkTimtU9AdSV7XoHKPKN6KT6B9VBso6TbsSmh0nEb6@mail.gmail.com>

I have to correct myself:

The latest 2.6 firmware with 675Xi phones can support SCA partially(The
other party can barge in calls initiated from Aastra phone, but Aastra phone
will drop any existing SCA calls when picking up)

On Tue, May 11, 2010 at 11:13 PM, Chris Chen <chris.chen2004 at gmail.com>wrote:

> I just tested the 2.6 firmware on my Aastra 55i, looks like they have made
> SCA working with my Polycom phone IP 650 now
> I will see if there any other problems with Aastra phone.
> Chris Chen
>
>
>
> On Tue, May 11, 2010 at 4:12 PM, Mark Maly <mark.maly at molcs.org> wrote:
>
>>  FYI-
>>
>>
>>
>> Got a ticket w/Aastra and they said,
>>
>>
>>
>> ?We have just released a new GA firmware that I would like for you to load
>> onto one of your phones to see if it resolves the issue. You can download
>> this from our website at www.aastratelecom.com/support and click on
>> Download Area and select the 6731i. The firmware version you are looking for
>> is 2.6 and it is listed under Current Software Release.
>>
>>
>>
>> Please load that on a test phone and let us know the results.?
>>
>>
>>
>> I?ll let you know,
>>
>> Mark
>>
>>
>>
>>
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West
>> *Sent:* Monday, May 03, 2010 8:00 PM
>>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Subject:* Re: [Freeswitch-users] Aastra and SCA
>>
>>
>>
>> Nope but we have exchanged some emails about it back and forth and some
>> beta firmware where they did half way fix it... but seems you need the
>> call-info header on ALL packets associated with the dialog of that call.
>>  They are still missing a few.
>>
>>
>>
>> /b
>>
>>
>>
>> On May 3, 2010, at 7:55 PM, Mark Maly wrote:
>>
>>
>>
>>   Brian,
>>
>>
>>
>> After my first reply I did receive a msg from Aastra support and whether
>> FS had reported the problem.   The contact wondered whether they [FS] had
>> been given any ?reference number? related to the problem.
>>
>>
>>
>> Would love to help and would pass ref num if you had one.
>>
>>
>>
>> Mark
>>
>>
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West
>> *Sent:* Sunday, May 02, 2010 11:14 PM
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Subject:* Re: [Freeswitch-users] Aastra and SCA
>>
>>
>>
>> Give up their exists a bug that prevents it from working on any aastra.
>>  Polycom and Cisco SPA work flawless.
>>
>>
>>
>> /b
>>
>> Sent from my iPad
>>
>>
>> On Apr 30, 2010, at 9:07 PM, "Mark Maly" <mark.maly at molcs.org> wrote:
>>
>>   Hi,
>>
>>
>>
>> I?ve tried to patiently figure this out by reading the wiki and this
>> list.  Unfortunately, I?ve been unable to get it right.
>>
>>
>>
>> I have 2 Aastra 6731is and a 51i and trying to get SCA working.  I?m
>> experiencing problems similar to the Cisco thread from last month ? outgoing
>> calls implement SCA well.  Incoming calls ring all lines and appearances
>> work, but when one phone is answered, the line appearances are removed from
>> the remaining phones.  I am not attempting to use any DNS.
>>
>>
>>
>> My configuration has all three phones plus FS on a local LAN.  Nothing too
>> fancy.  Each line is configured for Broadsoft SCA and SCA bridging  is
>> enabled globally for the phones.
>>
>>
>>
>> Trying to update/replace an old phone at my church.
>>
>>
>>
>> Any help would be greatly appreciated.
>>
>>
>>
>> Thanks, Mark
>>
>> Mark.maly at molcs.org
>>
>>
>>
>>   _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>  _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From david.ponzone at gmail.com  Tue May 11 22:46:57 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Wed, 12 May 2010 07:46:57 +0200
Subject: [Freeswitch-users] Freeswitch Scripting strangeness
In-Reply-To: <4BE9CE90.7050805@conspiracy.net>
References: <4BE9CE90.7050805@conspiracy.net>
Message-ID: <FD74BBBD-493D-48AC-B2E8-94B9D4D2029B@gmail.com>

Paul,

something is not clear for me in your description.
Is the script writing all logs for all calls to the same file, or to  
one file per call ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 11/05/2010 ? 23:39, paul bilke a ?crit :

> I am working on a system that invokes a LUA script of the dial plan  
> to handle interaction on the call.
> All is fine when running a single instance of the script (one  
> call).  The issue is when there is more than one call things become  
> erratic.
> Right now the script opens a log file who's name is based on the  
> date of the call and its DNIS using io.open with append.
> The script then writes lines to file as it runs detailing its  
> actions.  The same log function also uses posix.syslog to log the  
> same information
> via syslog.  All is fine when there is one call up.
> If there is more than one concurrent on two different DNIS's both  
> log files a present but have output that was created by the other  
> call.
> The syslog is sane but not the file of logging information.  The  
> time stamp will jump backward and pickup with information from the  
> other call
> (can tell via the UUID)
> It is my assumption that each call should spin up a thread and they  
> should share no data/variables/handles with other calls in the  
> system.  Looking at
> the logs indicates there is some issue but logging is not the only  
> issue since it seems that call events updating state information is  
> affecting the other
> call also.
> If my assumption on a isolated thread per call is incorrect let me  
> know.
> I inherited this code and suspect that running an application out of  
> the dialplan script is unconventional.
> Paul
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From b_ball_henry at hotmail.com  Tue May 11 22:48:14 2010
From: b_ball_henry at hotmail.com (Henry Huang)
Date: Wed, 12 May 2010 13:48:14 +0800
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <AANLkTikWXr8bdP2VV677-nhlZF7i2tLcv_wBzPRY--IF@mail.gmail.com>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<201005091701.03473.sos@sokhapkin.dyndns.org>
	<AANLkTimTdNs40gq6WTuvCtWdrVxgcyUYgsuYt4fcZZWe@mail.gmail.com>
	<201005101214.39428.sos@sokhapkin.dyndns.org>
	<AANLkTikWXr8bdP2VV677-nhlZF7i2tLcv_wBzPRY--IF@mail.gmail.com>
Message-ID: <AANLkTikrXS0qvTfdiN7hiOl627jznPXT6km-ETkZ0NKA@mail.gmail.com>

Sergey:

Did you tried the latest code of nibbliebill? and did it fix your problem?



On Tue, May 11, 2010 at 12:23 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> ok so you don't know for sure then, this is obviously a small box.
> My browser right now is using 500m to let me write this email to you.
>
> if you find nothing, it would suggest pools growing more than a real leak
> since valgrind would normally pick it up.
> I did not make this nibble bill module but I did correct the code in the
> place you pointed out.  Give it a try.
>
> I wish I could nibble-bill this mailing list =p
>
>
> On Mon, May 10, 2010 at 11:14 AM, Sergey Okhapkin <
> sos at sokhapkin.dyndns.org> wrote:
>
>> It's already on jira - FSMOD-48. Valgrind with --leak-check=full finds
>> nothing.
>>
>> Memory kept growing and growing, I had to restart FS when RSS reached 500M
>> to
>> avoid swap.
>>
>> On Monday 10 May 2010, Anthony Minessale wrote:
>> > Why does this thread continue on the list instead of being opened in
>> JIRA
>> > Can you see how this issue can be forgotten because it's on the mailing
>> >  list that gets 200 emails a day?
>> > When you show the code in the body of an email like that, then I have to
>> > spend a minuted wondering where in the code you were talking about
>> because
>> > it's not in the form of a unified diff.
>> >
>> > I appreciate that you do not normally have to deal with organized
>> software
>> > management but I must ask that you please spend a little time to learn
>> our
>> > customs and use JIRA and patches to communicate bug reports.  I am glad
>> you
>> > are helping to find problems but we really need to stay organized to
>> make
>> >  it forward.
>> >
>> > I redid the function in latest GIT.  It did look like it was doing a few
>> > things wrong but they probably would have surfaced in valgrind if you
>> were
>> > running it in full leak check mode.  When you say it goes to 300M, does
>> it
>> > keep going from there because there is a bit of memory that will be
>> pooled
>> > as you start to use more advanced features.  Did you ever watch it to
>> see
>> > how high it will go?
>> >
>> > On Sun, May 9, 2010 at 4:01 PM, Sergey Okhapkin
>> <sos at sokhapkin.dyndns.org>wrote:
>> > > Shouldn't label "end:" be BEFORE
>> > > switch_odbc_statement_handle_free(&stmt); ?
>> > > I think it should...
>> > >
>> > >
>> > > static switch_status_t bill_event(
>> > > ....
>> > >
>> > >       if (switch_odbc_handle_exec(globals.master_odbc, sql, &stmt,
>> NULL)
>> > > != SWITCH_ODBC_SUCCESS) {
>> > >                char *err_str;
>> > >                err_str =
>> > > switch_odbc_handle_get_error(globals.master_odbc, stmt);
>> > >                switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR,
>> > > "ERR:
>> > > [%s]\n[%s]\n", sql, switch_str_nil(err_str));
>> > >                switch_safe_free(err_str);
>> > >        } else {
>> > >                /* TODO: Failover to a flat/text file if DB is
>> unavailable
>> > > */
>> > >
>> > >                goto end;
>> > >        }
>> > >
>> > >        switch_odbc_statement_handle_free(&stmt);
>> > >
>> > > end:
>> > >
>> > > On Sunday 09 May 2010, Brian West wrote:
>> > > > What are you calling significant memory usage?
>> > > >
>> > > > /b
>> > > >
>> > > > On May 9, 2010, at 2:10 PM, Sergey Okhapkin wrote:
>> > > > > Valgrind output shows no significant leaks. This was discussed
>> > > > > already
>> > >
>> > > in
>> > >
>> > > > > this thread a month ago.
>> > > > >
>> > > > > On Sunday 09 May 2010, Brian West wrote:
>> > > > >> Please use valgrind and see where its leaking then open a jira.
>> > > > >>
>> > > > >> Nobody has been able to reproduce this in a lab nor provide any
>> > >
>> > > details
>> > >
>> > > > >> to assist in finding the issue... All I have seen is people
>> > >
>> > > complaining
>> > >
>> > > > >> about it and not doing what they should debugging the issue and
>> > > > >> reporting it.
>> > > > >>
>> > > > >> /b
>> > > >
>> > > > _______________________________________________
>> > > > FreeSWITCH-users mailing list
>> > > > FreeSWITCH-users at lists.freeswitch.org
>> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > > > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-user
>> > > >s http://www.freeswitch.org
>> > >
>> > > _______________________________________________
>> > > FreeSWITCH-users mailing list
>> > > FreeSWITCH-users at lists.freeswitch.org
>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > > http://www.freeswitch.org
>> >
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From jan.berger at video24.no  Wed May 12 02:36:54 2010
From: jan.berger at video24.no (Jan Berger)
Date: Wed, 12 May 2010 11:36:54 +0200
Subject: [Freeswitch-users] SIPfoundry ACD w/FreeSWITCH
Message-ID: <9600A2F5648241FFA8C1CB1CDF89DB1F@dell9400>

Hi,

 

Maybe old news, but found this design page with reference to FreeSWITCH on
SIPfoundry. http://sipx-wiki.calivia.com/index.php/ACD_Proposed_Architecture


 

They seem to be increasingly found of FreeSWITCH.  

 

Jan

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From sos at sokhapkin.dyndns.org  Wed May 12 03:23:50 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 12 May 2010 06:23:50 -0400
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <AANLkTikrXS0qvTfdiN7hiOl627jznPXT6km-ETkZ0NKA@mail.gmail.com>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<AANLkTikWXr8bdP2VV677-nhlZF7i2tLcv_wBzPRY--IF@mail.gmail.com>
	<AANLkTikrXS0qvTfdiN7hiOl627jznPXT6km-ETkZ0NKA@mail.gmail.com>
Message-ID: <201005120623.50868.sos@sokhapkin.dyndns.org>

I didn't try yesterday Anthony's update, but his previous fix works OK.

Please note that comments in nibblebill config file and default actions in 
nibblebill code for lowbal_action, nobal_action and percall_action are 
incorrect, the parameters are not the applications to execute, but dialplan 
extensions to transfer the call to.

On Wednesday 12 May 2010, Henry Huang wrote:
> Sergey:
> 
> Did you tried the latest code of nibbliebill? and did it fix your problem?
> 
> 
> 
> On Tue, May 11, 2010 at 12:23 AM, Anthony Minessale <
> 
> anthony.minessale at gmail.com> wrote:
> > ok so you don't know for sure then, this is obviously a small box.
> > My browser right now is using 500m to let me write this email to you.
> >
> > if you find nothing, it would suggest pools growing more than a real leak
> > since valgrind would normally pick it up.
> > I did not make this nibble bill module but I did correct the code in the
> > place you pointed out.  Give it a try.
> >
> > I wish I could nibble-bill this mailing list =p
> >
> >
> > On Mon, May 10, 2010 at 11:14 AM, Sergey Okhapkin <
> >
> > sos at sokhapkin.dyndns.org> wrote:
> >> It's already on jira - FSMOD-48. Valgrind with --leak-check=full finds
> >> nothing.
> >>
> >> Memory kept growing and growing, I had to restart FS when RSS reached
> >> 500M to
> >> avoid swap.
> >>
> >> On Monday 10 May 2010, Anthony Minessale wrote:
> >> > Why does this thread continue on the list instead of being opened in
> >>
> >> JIRA
> >>
> >> > Can you see how this issue can be forgotten because it's on the
> >> > mailing list that gets 200 emails a day?
> >> > When you show the code in the body of an email like that, then I have
> >> > to spend a minuted wondering where in the code you were talking about
> >>
> >> because
> >>
> >> > it's not in the form of a unified diff.
> >> >
> >> > I appreciate that you do not normally have to deal with organized
> >>
> >> software
> >>
> >> > management but I must ask that you please spend a little time to learn
> >>
> >> our
> >>
> >> > customs and use JIRA and patches to communicate bug reports.  I am
> >> > glad
> >>
> >> you
> >>
> >> > are helping to find problems but we really need to stay organized to
> >>
> >> make
> >>
> >> >  it forward.
> >> >
> >> > I redid the function in latest GIT.  It did look like it was doing a
> >> > few things wrong but they probably would have surfaced in valgrind if
> >> > you
> >>
> >> were
> >>
> >> > running it in full leak check mode.  When you say it goes to 300M,
> >> > does
> >>
> >> it
> >>
> >> > keep going from there because there is a bit of memory that will be
> >>
> >> pooled
> >>
> >> > as you start to use more advanced features.  Did you ever watch it to
> >>
> >> see
> >>
> >> > how high it will go?
> >> >
> >> > On Sun, May 9, 2010 at 4:01 PM, Sergey Okhapkin
> >>
> >> <sos at sokhapkin.dyndns.org>wrote:
> >> > > Shouldn't label "end:" be BEFORE
> >> > > switch_odbc_statement_handle_free(&stmt); ?
> >> > > I think it should...
> >> > >
> >> > >
> >> > > static switch_status_t bill_event(
> >> > > ....
> >> > >
> >> > >       if (switch_odbc_handle_exec(globals.master_odbc, sql, &stmt,
> >>
> >> NULL)
> >>
> >> > > != SWITCH_ODBC_SUCCESS) {
> >> > >                char *err_str;
> >> > >                err_str =
> >> > > switch_odbc_handle_get_error(globals.master_odbc, stmt);
> >> > >                switch_log_printf(SWITCH_CHANNEL_LOG,
> >> > > SWITCH_LOG_ERROR, "ERR:
> >> > > [%s]\n[%s]\n", sql, switch_str_nil(err_str));
> >> > >                switch_safe_free(err_str);
> >> > >        } else {
> >> > >                /* TODO: Failover to a flat/text file if DB is
> >>
> >> unavailable
> >>
> >> > > */
> >> > >
> >> > >                goto end;
> >> > >        }
> >> > >
> >> > >        switch_odbc_statement_handle_free(&stmt);
> >> > >
> >> > > end:
> >> > >
> >> > > On Sunday 09 May 2010, Brian West wrote:
> >> > > > What are you calling significant memory usage?
> >> > > >
> >> > > > /b
> >> > > >
> >> > > > On May 9, 2010, at 2:10 PM, Sergey Okhapkin wrote:
> >> > > > > Valgrind output shows no significant leaks. This was discussed
> >> > > > > already
> >> > >
> >> > > in
> >> > >
> >> > > > > this thread a month ago.
> >> > > > >
> >> > > > > On Sunday 09 May 2010, Brian West wrote:
> >> > > > >> Please use valgrind and see where its leaking then open a jira.
> >> > > > >>
> >> > > > >> Nobody has been able to reproduce this in a lab nor provide any
> >> > >
> >> > > details
> >> > >
> >> > > > >> to assist in finding the issue... All I have seen is people
> >> > >
> >> > > complaining
> >> > >
> >> > > > >> about it and not doing what they should debugging the issue and
> >> > > > >> reporting it.
> >> > > > >>
> >> > > > >> /b
> >> > > >
> >> > > > _______________________________________________
> >> > > > FreeSWITCH-users mailing list
> >> > > > FreeSWITCH-users at lists.freeswitch.org
> >> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > > > UNSUBSCRIBE:
> >>
> >> http://lists.freeswitch.org/mailman/options/freeswitch-user
> >>
> >> > > >s http://www.freeswitch.org
> >> > >
> >> > > _______________________________________________
> >> > > FreeSWITCH-users mailing list
> >> > > FreeSWITCH-users at lists.freeswitch.org
> >> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > > UNSUBSCRIBE:
> >>
> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>
> >> > > http://www.freeswitch.org
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > 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.minessal
> >e 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 con
> >ference.freeswitch.org> pstn:+19193869900
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 



From b_ball_henry at hotmail.com  Wed May 12 03:47:57 2010
From: b_ball_henry at hotmail.com (Henry Huang)
Date: Wed, 12 May 2010 18:47:57 +0800
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <201005120623.50868.sos@sokhapkin.dyndns.org>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<AANLkTikWXr8bdP2VV677-nhlZF7i2tLcv_wBzPRY--IF@mail.gmail.com>
	<AANLkTikrXS0qvTfdiN7hiOl627jznPXT6km-ETkZ0NKA@mail.gmail.com>
	<201005120623.50868.sos@sokhapkin.dyndns.org>
Message-ID: <AANLkTilzor0UZG3X6XsltRs_uBGkR3aqmOSmhHZRIQvS@mail.gmail.com>

Sergey:

just making sure. When you say it works okay, did you mean the memory leak
stopped?



On Wed, May 12, 2010 at 6:23 PM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> I didn't try yesterday Anthony's update, but his previous fix works OK.
>
> Please note that comments in nibblebill config file and default actions in
> nibblebill code for lowbal_action, nobal_action and percall_action are
> incorrect, the parameters are not the applications to execute, but dialplan
> extensions to transfer the call to.
>
> On Wednesday 12 May 2010, Henry Huang wrote:
> > Sergey:
> >
> > Did you tried the latest code of nibbliebill? and did it fix your
> problem?
> >
> >
> >
> > On Tue, May 11, 2010 at 12:23 AM, Anthony Minessale <
> >
> > anthony.minessale at gmail.com> wrote:
> > > ok so you don't know for sure then, this is obviously a small box.
> > > My browser right now is using 500m to let me write this email to you.
> > >
> > > if you find nothing, it would suggest pools growing more than a real
> leak
> > > since valgrind would normally pick it up.
> > > I did not make this nibble bill module but I did correct the code in
> the
> > > place you pointed out.  Give it a try.
> > >
> > > I wish I could nibble-bill this mailing list =p
> > >
> > >
> > > On Mon, May 10, 2010 at 11:14 AM, Sergey Okhapkin <
> > >
> > > sos at sokhapkin.dyndns.org> wrote:
> > >> It's already on jira - FSMOD-48. Valgrind with --leak-check=full finds
> > >> nothing.
> > >>
> > >> Memory kept growing and growing, I had to restart FS when RSS reached
> > >> 500M to
> > >> avoid swap.
> > >>
> > >> On Monday 10 May 2010, Anthony Minessale wrote:
> > >> > Why does this thread continue on the list instead of being opened in
> > >>
> > >> JIRA
> > >>
> > >> > Can you see how this issue can be forgotten because it's on the
> > >> > mailing list that gets 200 emails a day?
> > >> > When you show the code in the body of an email like that, then I
> have
> > >> > to spend a minuted wondering where in the code you were talking
> about
> > >>
> > >> because
> > >>
> > >> > it's not in the form of a unified diff.
> > >> >
> > >> > I appreciate that you do not normally have to deal with organized
> > >>
> > >> software
> > >>
> > >> > management but I must ask that you please spend a little time to
> learn
> > >>
> > >> our
> > >>
> > >> > customs and use JIRA and patches to communicate bug reports.  I am
> > >> > glad
> > >>
> > >> you
> > >>
> > >> > are helping to find problems but we really need to stay organized to
> > >>
> > >> make
> > >>
> > >> >  it forward.
> > >> >
> > >> > I redid the function in latest GIT.  It did look like it was doing a
> > >> > few things wrong but they probably would have surfaced in valgrind
> if
> > >> > you
> > >>
> > >> were
> > >>
> > >> > running it in full leak check mode.  When you say it goes to 300M,
> > >> > does
> > >>
> > >> it
> > >>
> > >> > keep going from there because there is a bit of memory that will be
> > >>
> > >> pooled
> > >>
> > >> > as you start to use more advanced features.  Did you ever watch it
> to
> > >>
> > >> see
> > >>
> > >> > how high it will go?
> > >> >
> > >> > On Sun, May 9, 2010 at 4:01 PM, Sergey Okhapkin
> > >>
> > >> <sos at sokhapkin.dyndns.org>wrote:
> > >> > > Shouldn't label "end:" be BEFORE
> > >> > > switch_odbc_statement_handle_free(&stmt); ?
> > >> > > I think it should...
> > >> > >
> > >> > >
> > >> > > static switch_status_t bill_event(
> > >> > > ....
> > >> > >
> > >> > >       if (switch_odbc_handle_exec(globals.master_odbc, sql, &stmt,
> > >>
> > >> NULL)
> > >>
> > >> > > != SWITCH_ODBC_SUCCESS) {
> > >> > >                char *err_str;
> > >> > >                err_str =
> > >> > > switch_odbc_handle_get_error(globals.master_odbc, stmt);
> > >> > >                switch_log_printf(SWITCH_CHANNEL_LOG,
> > >> > > SWITCH_LOG_ERROR, "ERR:
> > >> > > [%s]\n[%s]\n", sql, switch_str_nil(err_str));
> > >> > >                switch_safe_free(err_str);
> > >> > >        } else {
> > >> > >                /* TODO: Failover to a flat/text file if DB is
> > >>
> > >> unavailable
> > >>
> > >> > > */
> > >> > >
> > >> > >                goto end;
> > >> > >        }
> > >> > >
> > >> > >        switch_odbc_statement_handle_free(&stmt);
> > >> > >
> > >> > > end:
> > >> > >
> > >> > > On Sunday 09 May 2010, Brian West wrote:
> > >> > > > What are you calling significant memory usage?
> > >> > > >
> > >> > > > /b
> > >> > > >
> > >> > > > On May 9, 2010, at 2:10 PM, Sergey Okhapkin wrote:
> > >> > > > > Valgrind output shows no significant leaks. This was discussed
> > >> > > > > already
> > >> > >
> > >> > > in
> > >> > >
> > >> > > > > this thread a month ago.
> > >> > > > >
> > >> > > > > On Sunday 09 May 2010, Brian West wrote:
> > >> > > > >> Please use valgrind and see where its leaking then open a
> jira.
> > >> > > > >>
> > >> > > > >> Nobody has been able to reproduce this in a lab nor provide
> any
> > >> > >
> > >> > > details
> > >> > >
> > >> > > > >> to assist in finding the issue... All I have seen is people
> > >> > >
> > >> > > complaining
> > >> > >
> > >> > > > >> about it and not doing what they should debugging the issue
> and
> > >> > > > >> reporting it.
> > >> > > > >>
> > >> > > > >> /b
> > >> > > >
> > >> > > > _______________________________________________
> > >> > > > FreeSWITCH-users mailing list
> > >> > > > FreeSWITCH-users at lists.freeswitch.org
> > >> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> > > > UNSUBSCRIBE:
> > >>
> > >> http://lists.freeswitch.org/mailman/options/freeswitch-user
> > >>
> > >> > > >s http://www.freeswitch.org
> > >> > >
> > >> > > _______________________________________________
> > >> > > FreeSWITCH-users mailing list
> > >> > > FreeSWITCH-users at lists.freeswitch.org
> > >> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> > > UNSUBSCRIBE:
> > >>
> > >> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >>
> > >> > > http://www.freeswitch.org
> > >>
> > >> _______________________________________________
> > >> FreeSWITCH-users mailing list
> > >> FreeSWITCH-users at lists.freeswitch.org
> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >> http://www.freeswitch.org
> > >
> > > --
> > > Anthony Minessale II
> > >
> > > FreeSWITCH http://www.freeswitch.org/
> > > ClueCon http://www.cluecon.com/
> > > Twitter: http://twitter.com/FreeSWITCH_wire
> > >
> > > AIM: anthm
> > > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com><
> MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> <PAYPAL%3Aanthony.minessal
> > >e at gmail.com> IRC: irc.freenode.net #freeswitch
> > >
> > > FreeSWITCH Developer Conference
> > > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org><
> sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> > > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> <googletalk%3Aconf%2B888 at con
> > >ference.freeswitch.org> pstn:+19193869900
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From sos at sokhapkin.dyndns.org  Wed May 12 03:59:04 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 12 May 2010 06:59:04 -0400
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <AANLkTilzor0UZG3X6XsltRs_uBGkR3aqmOSmhHZRIQvS@mail.gmail.com>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<201005120623.50868.sos@sokhapkin.dyndns.org>
	<AANLkTilzor0UZG3X6XsltRs_uBGkR3aqmOSmhHZRIQvS@mail.gmail.com>
Message-ID: <201005120659.04426.sos@sokhapkin.dyndns.org>

Correct, no more unbouded memory growth.

On Wednesday 12 May 2010, Henry Huang wrote:
> Sergey:
> 
> just making sure. When you say it works okay, did you mean the memory leak
> stopped?
> 
> 
> 
> On Wed, May 12, 2010 at 6:23 PM, Sergey Okhapkin
> 
> <sos at sokhapkin.dyndns.org>wrote:
> > I didn't try yesterday Anthony's update, but his previous fix works OK.
> >
> > Please note that comments in nibblebill config file and default actions
> > in nibblebill code for lowbal_action, nobal_action and percall_action are
> > incorrect, the parameters are not the applications to execute, but
> > dialplan extensions to transfer the call to.
> >
> > On Wednesday 12 May 2010, Henry Huang wrote:
> > > Sergey:
> > >
> > > Did you tried the latest code of nibbliebill? and did it fix your
> >
> > problem?
> >
> > > On Tue, May 11, 2010 at 12:23 AM, Anthony Minessale <
> > >
> > > anthony.minessale at gmail.com> wrote:
> > > > ok so you don't know for sure then, this is obviously a small box.
> > > > My browser right now is using 500m to let me write this email to you.
> > > >
> > > > if you find nothing, it would suggest pools growing more than a real
> >
> > leak
> >
> > > > since valgrind would normally pick it up.
> > > > I did not make this nibble bill module but I did correct the code in
> >
> > the
> >
> > > > place you pointed out.  Give it a try.
> > > >
> > > > I wish I could nibble-bill this mailing list =p
> > > >
> > > >
> > > > On Mon, May 10, 2010 at 11:14 AM, Sergey Okhapkin <
> > > >
> > > > sos at sokhapkin.dyndns.org> wrote:
> > > >> It's already on jira - FSMOD-48. Valgrind with --leak-check=full
> > > >> finds nothing.
> > > >>
> > > >> Memory kept growing and growing, I had to restart FS when RSS
> > > >> reached 500M to
> > > >> avoid swap.
> > > >>
> > > >> On Monday 10 May 2010, Anthony Minessale wrote:
> > > >> > Why does this thread continue on the list instead of being opened
> > > >> > in
> > > >>
> > > >> JIRA
> > > >>
> > > >> > Can you see how this issue can be forgotten because it's on the
> > > >> > mailing list that gets 200 emails a day?
> > > >> > When you show the code in the body of an email like that, then I
> >
> > have
> >
> > > >> > to spend a minuted wondering where in the code you were talking
> >
> > about
> >
> > > >> because
> > > >>
> > > >> > it's not in the form of a unified diff.
> > > >> >
> > > >> > I appreciate that you do not normally have to deal with organized
> > > >>
> > > >> software
> > > >>
> > > >> > management but I must ask that you please spend a little time to
> >
> > learn
> >
> > > >> our
> > > >>
> > > >> > customs and use JIRA and patches to communicate bug reports.  I am
> > > >> > glad
> > > >>
> > > >> you
> > > >>
> > > >> > are helping to find problems but we really need to stay organized
> > > >> > to
> > > >>
> > > >> make
> > > >>
> > > >> >  it forward.
> > > >> >
> > > >> > I redid the function in latest GIT.  It did look like it was doing
> > > >> > a few things wrong but they probably would have surfaced in
> > > >> > valgrind
> >
> > if
> >
> > > >> > you
> > > >>
> > > >> were
> > > >>
> > > >> > running it in full leak check mode.  When you say it goes to 300M,
> > > >> > does
> > > >>
> > > >> it
> > > >>
> > > >> > keep going from there because there is a bit of memory that will
> > > >> > be
> > > >>
> > > >> pooled
> > > >>
> > > >> > as you start to use more advanced features.  Did you ever watch it
> >
> > to
> >
> > > >> see
> > > >>
> > > >> > how high it will go?
> > > >> >
> > > >> > On Sun, May 9, 2010 at 4:01 PM, Sergey Okhapkin
> > > >>
> > > >> <sos at sokhapkin.dyndns.org>wrote:
> > > >> > > Shouldn't label "end:" be BEFORE
> > > >> > > switch_odbc_statement_handle_free(&stmt); ?
> > > >> > > I think it should...
> > > >> > >
> > > >> > >
> > > >> > > static switch_status_t bill_event(
> > > >> > > ....
> > > >> > >
> > > >> > >       if (switch_odbc_handle_exec(globals.master_odbc, sql,
> > > >> > > &stmt,
> > > >>
> > > >> NULL)
> > > >>
> > > >> > > != SWITCH_ODBC_SUCCESS) {
> > > >> > >                char *err_str;
> > > >> > >                err_str =
> > > >> > > switch_odbc_handle_get_error(globals.master_odbc, stmt);
> > > >> > >                switch_log_printf(SWITCH_CHANNEL_LOG,
> > > >> > > SWITCH_LOG_ERROR, "ERR:
> > > >> > > [%s]\n[%s]\n", sql, switch_str_nil(err_str));
> > > >> > >                switch_safe_free(err_str);
> > > >> > >        } else {
> > > >> > >                /* TODO: Failover to a flat/text file if DB is
> > > >>
> > > >> unavailable
> > > >>
> > > >> > > */
> > > >> > >
> > > >> > >                goto end;
> > > >> > >        }
> > > >> > >
> > > >> > >        switch_odbc_statement_handle_free(&stmt);
> > > >> > >
> > > >> > > end:
> > > >> > >
> > > >> > > On Sunday 09 May 2010, Brian West wrote:
> > > >> > > > What are you calling significant memory usage?
> > > >> > > >
> > > >> > > > /b
> > > >> > > >
> > > >> > > > On May 9, 2010, at 2:10 PM, Sergey Okhapkin wrote:
> > > >> > > > > Valgrind output shows no significant leaks. This was
> > > >> > > > > discussed already
> > > >> > >
> > > >> > > in
> > > >> > >
> > > >> > > > > this thread a month ago.
> > > >> > > > >
> > > >> > > > > On Sunday 09 May 2010, Brian West wrote:
> > > >> > > > >> Please use valgrind and see where its leaking then open a
> >
> > jira.
> >
> > > >> > > > >> Nobody has been able to reproduce this in a lab nor provide
> >
> > any
> >
> > > >> > > details
> > > >> > >
> > > >> > > > >> to assist in finding the issue... All I have seen is people
> > > >> > >
> > > >> > > complaining
> > > >> > >
> > > >> > > > >> about it and not doing what they should debugging the issue
> >
> > and
> >
> > > >> > > > >> reporting it.
> > > >> > > > >>
> > > >> > > > >> /b
> > > >> > > >
> > > >> > > > _______________________________________________
> > > >> > > > FreeSWITCH-users mailing list
> > > >> > > > FreeSWITCH-users at lists.freeswitch.org
> > > >> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > >> > > > UNSUBSCRIBE:
> > > >>
> > > >> http://lists.freeswitch.org/mailman/options/freeswitch-user
> > > >>
> > > >> > > >s http://www.freeswitch.org
> > > >> > >
> > > >> > > _______________________________________________
> > > >> > > FreeSWITCH-users mailing list
> > > >> > > FreeSWITCH-users at lists.freeswitch.org
> > > >> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > >> > > UNSUBSCRIBE:
> > > >>
> > > >> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > >>
> > > >> > > http://www.freeswitch.org
> > > >>
> > > >> _______________________________________________
> > > >> FreeSWITCH-users mailing list
> > > >> FreeSWITCH-users at lists.freeswitch.org
> > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > >> UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > > >> http://www.freeswitch.org
> > > >
> > > > --
> > > > Anthony Minessale II
> > > >
> > > > FreeSWITCH http://www.freeswitch.org/
> > > > ClueCon http://www.cluecon.com/
> > > > Twitter: http://twitter.com/FreeSWITCH_wire
> > > >
> > > > AIM: anthm
> > > > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com
> > > >><
> >
> > MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com
> >>
> >
> > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.mine
> > > >ssale at gmail.com>
> >
> > <PAYPAL%3Aanthony.minessal
> >
> > > >e at gmail.com> IRC: irc.freenode.net #freeswitch
> > > >
> > > > FreeSWITCH Developer Conference
> > > > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org
> > > >><
> >
> > sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org
> >>
> >
> > > > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888
> > > >@conference.freeswitch.org>
> >
> > <googletalk%3Aconf%2B888 at con
> >
> > > >ference.freeswitch.org> pstn:+19193869900
> > > >
> > > > _______________________________________________
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > UNSUBSCRIBE:
> >
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> >
> > > > http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 



From shaheryarkh at googlemail.com  Wed May 12 04:48:00 2010
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Wed, 12 May 2010 16:48:00 +0500
Subject: [Freeswitch-users] FreeSWITCH startup failure
Message-ID: <AANLkTinuEQVMWopPUrCYJSQa2XHaMHVASI882_aH8k71@mail.gmail.com>

Hi,

For last two days i am trying to setup a freeswitch box for an voip
application testing against it. FreeSWITCH compiles successfully but when i
try to start it i get a lot critical errors from switch_core_sqldb.c, e.g.


2010-05-12 16:35:42.691358 [ERR] switch_core_sqldb.c:404 SQL ERR [no such
table: interfaces]
insert into interfaces (type,name,description,syntax,ikey,filename,hostname)
values('chat','event','','','mod_dptools','/usr/local/freeswitch/mod/mod_dptools.so','lenovo');
insert into interfaces (type,name,description,syntax,ikey,filename,hostname)
values('chat','api','','','mod_dptools','/usr/local/freeswitch/mod/mod_dptools.so','lenovo');
insert into interfaces (type,name,description,syntax,ikey,filename,hostname)
values('api','expr','Eval an
expression','<expr>','mod_expr','/usr/local/freeswitch/mod/mod_expr.so','lenovo');

2010-05-12 16:35:42.691419 [ERR] switch_core_sqldb.c:670 SQL ERR [no such
table: interfaces]
2010-05-12 16:35:42.791490 [CRIT] switch_core_sqldb.c:889 SQL thread unable
to commit transaction, records lost!

and it continues till my machine eventually hangs. Its happening with latest
git revision, i don't know how to get revision number in git (i am not much
familiar with it) but "git log" command give this as first line.

commit 5530de66ef6880621086c6276731504459eb6709
Author: Rupa Schomaker <rupa at rupa.com>
Date:   Wed May 12 03:32:18 2010 -0500

I suspect the last line, i.e.
2010-05-12 16:35:42.791490 [CRIT] switch_core_sqldb.c:889 SQL thread unable
to commit transaction, records lost!

is the reason for these errors but i am not sure why its happening.

Thank you.


-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From babak.freeswitch at gmail.com  Wed May 12 06:22:49 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Wed, 12 May 2010 17:52:49 +0430
Subject: [Freeswitch-users] build error on latest git
Message-ID: <AANLkTikcvi3S-mW7IJR_bU4v9gwmdTwECmRypMjX-G3J@mail.gmail.com>

hi
I got this error on latest git pulled today:
Error 1 error LNK2019: unresolved external symbol _saturate16 referenced in
function _sig_tone_rx at 12 sig_tone.obj
Error 2 fatal error LNK1120: 1 unresolved externals
D:\freeswitch\Debug\libspandsp.dll
thanks
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From rupa at rupa.com  Wed May 12 06:49:27 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 12 May 2010 08:49:27 -0500
Subject: [Freeswitch-users] FreeSWITCH startup failure
In-Reply-To: <AANLkTinuEQVMWopPUrCYJSQa2XHaMHVASI882_aH8k71@mail.gmail.com>
References: <AANLkTinuEQVMWopPUrCYJSQa2XHaMHVASI882_aH8k71@mail.gmail.com>
Message-ID: <AANLkTimOPHv6StrrobNll6HbzV_FqdQrLE1i6zidqiW2@mail.gmail.com>

THat commit just removes a .rej file, don't see how it can affect

$ git diff
--stat 1d2c64d33bf033788c5521c212b861ba11c229af..5530de66ef6880621086c6276731504459eb6709

(those are the two SHA1s I'm getting stats on,  first is tony's just before
my changes, second is mine)

 src/include/switch_types.h.orig                    | 1714 ------
 .../mod_conference/mod_conference.c.orig           | 6343
--------------------
 .../mod_conference/mod_conference.c.rej            |   50 -
 src/mod/formats/mod_sndfile/mod_sndfile.c.orig     |  462 --
 .../languages/mod_python/mod_python_wrap.cpp.rej   |   38 -
 src/switch_core.c.orig                             | 2018 -------
 src/switch_time.c.orig                             | 1962 ------
 7 files changed, 0 insertions(+), 12587 deletions(-)


Don't see how removing patch droppings can possibly cause your problem.

The real issue is that the interfaces table doesn't exist.  The last error
is because the table didn't exist for the records to be inserted into.  As
with all core tables, this SHOULD be auto-created.

Are you running against odbc for core or sqlite?

On Wed, May 12, 2010 at 6:48 AM, Muhammad Shahzad <
shaheryarkh at googlemail.com> wrote:

> Hi,
>
> For last two days i am trying to setup a freeswitch box for an voip
> application testing against it. FreeSWITCH compiles successfully but when i
> try to start it i get a lot critical errors from switch_core_sqldb.c, e.g.
>
>
> 2010-05-12 16:35:42.691358 [ERR] switch_core_sqldb.c:404 SQL ERR [no such
> table: interfaces]
> insert into interfaces
> (type,name,description,syntax,ikey,filename,hostname)
> values('chat','event','','','mod_dptools','/usr/local/freeswitch/mod/mod_dptools.so','lenovo');
> insert into interfaces
> (type,name,description,syntax,ikey,filename,hostname)
> values('chat','api','','','mod_dptools','/usr/local/freeswitch/mod/mod_dptools.so','lenovo');
> insert into interfaces
> (type,name,description,syntax,ikey,filename,hostname)
> values('api','expr','Eval an
> expression','<expr>','mod_expr','/usr/local/freeswitch/mod/mod_expr.so','lenovo');
>
> 2010-05-12 16:35:42.691419 [ERR] switch_core_sqldb.c:670 SQL ERR [no such
> table: interfaces]
> 2010-05-12 16:35:42.791490 [CRIT] switch_core_sqldb.c:889 SQL thread unable
> to commit transaction, records lost!
>
> and it continues till my machine eventually hangs. Its happening with
> latest git revision, i don't know how to get revision number in git (i am
> not much familiar with it) but "git log" command give this as first line.
>
> commit 5530de66ef6880621086c6276731504459eb6709
> Author: Rupa Schomaker <rupa at rupa.com>
> Date:   Wed May 12 03:32:18 2010 -0500
>
> I suspect the last line, i.e.
> 2010-05-12 16:35:42.791490 [CRIT] switch_core_sqldb.c:889 SQL thread unable
> to commit transaction, records lost!
>
> is the reason for these errors but i am not sure why its happening.
>
> Thank you.
>
>
> --
> Muhammad Shahzad
> -----------------------------------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +92 334 422 40 88
> MSN: shari_786pk at hotmail.com
> Email: shaheryarkh at googlemail.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From woof at iwoof.org  Wed May 12 09:05:29 2010
From: woof at iwoof.org (Andy Spitzer)
Date: Wed, 12 May 2010 12:05:29 -0400
Subject: [Freeswitch-users] SIPfoundry ACD w/FreeSWITCH
In-Reply-To: <9600A2F5648241FFA8C1CB1CDF89DB1F@dell9400>
References: <9600A2F5648241FFA8C1CB1CDF89DB1F@dell9400>
Message-ID: <AANLkTineZINUnVeSkgHhOW85RJfLA-uWnK8CTwrzCkaM@mail.gmail.com>

Woof!

On Wed, May 12, 2010 at 5:36 AM, Jan Berger <jan.berger at video24.no> wrote:
> Hi,
> Maybe old news, but found this design page with reference to FreeSWITCH on
> SIPfoundry. http://sipx-wiki.calivia.com/index.php/ACD_Proposed_Architecture

Nothing (visible) has come of that effort.  Since Avaya has taken over
Nortel, which took over Pingtel, some of the features of sipXecs have
been removed from open source in order to prop up interest in the
commercial version.  It isn't clear, but perhaps the FreeSWITCH based
ACD is part of that hold back.

--Woof!


From shroukkhan at softverk.is  Wed May 12 09:27:32 2010
From: shroukkhan at softverk.is (Shrouk Khan)
Date: Wed, 12 May 2010 23:27:32 +0700
Subject: [Freeswitch-users] files.freeswitch.org is down !
Message-ID: <AANLkTilONLE3bFww5hUUJdEJfyP0vG2OC4dZ-jesbIc1@mail.gmail.com>

hi ,
was trying to compile and install freeswitch 1.0.6 and it keeps getting
stuck at make stage:

making install mod_curl
--2010-05-12 23:13:27--
http://files.freeswitch.org/downloads/libs/json-c-0.9.tar.gz
Resolving files.freeswitch.org... 69.174.57.101
Connecting to files.freeswitch.org|69.174.57.101|:80... failed: Connection
refused.
cannot find json-c-0.9.tar.gz
make[5]: *** [/usr/src/freeswitch-1.0.6/libs/json-c-0.9] Error 1
make[4]: *** [install] Error 1
make[3]: *** [mod_curl-install] Error 1
make[2]: *** [install-recursive] Error 1


i tried browsing to that site using the browser and that didnt work either.

is the server down ? when will it be up ?

-- 
Regards

Shrouk Khan (Khan)
System Administrator / Telecommunication System Developer
Office:   +354 4400807 (Reykjavik)
          +44   2031370800 (London)
Mobile:  +66  875049439 (Bangkok)

Web: www.softverk.is

Reykjavik, Iceland  //  London, UK   //  Bangkok, Thailand
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From anthony.minessale at gmail.com  Wed May 12 09:32:33 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 12 May 2010 11:32:33 -0500
Subject: [Freeswitch-users] SIPfoundry ACD w/FreeSWITCH
In-Reply-To: <AANLkTineZINUnVeSkgHhOW85RJfLA-uWnK8CTwrzCkaM@mail.gmail.com>
References: <9600A2F5648241FFA8C1CB1CDF89DB1F@dell9400>
	<AANLkTineZINUnVeSkgHhOW85RJfLA-uWnK8CTwrzCkaM@mail.gmail.com>
Message-ID: <AANLkTim-T1UlDOhLR43M98QPmj_bsOBSYtE90JfI9M6a@mail.gmail.com>

Well you can take the FreeSWITCH out of the free version but you can't take
the Free out of FreeSWITCH ;)



On Wed, May 12, 2010 at 11:05 AM, Andy Spitzer <woof at iwoof.org> wrote:

> Woof!
>
> On Wed, May 12, 2010 at 5:36 AM, Jan Berger <jan.berger at video24.no> wrote:
> > Hi,
> > Maybe old news, but found this design page with reference to FreeSWITCH
> on
> > SIPfoundry.
> http://sipx-wiki.calivia.com/index.php/ACD_Proposed_Architecture
>
> Nothing (visible) has come of that effort.  Since Avaya has taken over
> Nortel, which took over Pingtel, some of the features of sipXecs have
> been removed from open source in order to prop up interest in the
> commercial version.  It isn't clear, but perhaps the FreeSWITCH based
> ACD is part of that hold back.
>
> --Woof!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
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
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From brian at freeswitch.org  Wed May 12 09:33:39 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 12 May 2010 11:33:39 -0500
Subject: [Freeswitch-users] files.freeswitch.org is down !
In-Reply-To: <AANLkTilONLE3bFww5hUUJdEJfyP0vG2OC4dZ-jesbIc1@mail.gmail.com>
References: <AANLkTilONLE3bFww5hUUJdEJfyP0vG2OC4dZ-jesbIc1@mail.gmail.com>
Message-ID: <6E84C50B-EDED-419C-9E75-6E31E08BD103@freeswitch.org>

Sounds like you have a firewall thats blocking it.

/b


On May 12, 2010, at 11:27 AM, Shrouk Khan wrote:

> hi ,
> was trying to compile and install freeswitch 1.0.6 and it keeps getting stuck at make stage:
> 
> making install mod_curl
> --2010-05-12 23:13:27--  http://files.freeswitch.org/downloads/libs/json-c-0.9.tar.gz
> Resolving files.freeswitch.org... 69.174.57.101
> Connecting to files.freeswitch.org|69.174.57.101|:80... failed: Connection refused.
> cannot find json-c-0.9.tar.gz
> make[5]: *** [/usr/src/freeswitch-1.0.6/libs/json-c-0.9] Error 1
> make[4]: *** [install] Error 1
> make[3]: *** [mod_curl-install] Error 1
> make[2]: *** [install-recursive] Error 1
> 
> 
> i tried browsing to that site using the browser and that didnt work either.
> 
> is the server down ? when will it be up ? 
> 
> -- 
> Regards
> 
> Shrouk Khan (Khan)
> System Administrator / Telecommunication System Developer
> Office:   +354 4400807 (Reykjavik)
>           +44   2031370800 (London)
> Mobile:  +66  875049439 (Bangkok)
> 
> Web: www.softverk.is
> 
> Reykjavik, Iceland  //  London, UK   //  Bangkok, Thailand
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Wed May 12 09:34:15 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 12 May 2010 11:34:15 -0500
Subject: [Freeswitch-users] files.freeswitch.org is down !
In-Reply-To: <AANLkTilONLE3bFww5hUUJdEJfyP0vG2OC4dZ-jesbIc1@mail.gmail.com>
References: <AANLkTilONLE3bFww5hUUJdEJfyP0vG2OC4dZ-jesbIc1@mail.gmail.com>
Message-ID: <55DE6C2F-9D59-4B36-9D7E-96B5FE48A1DA@freeswitch.org>

<2>:wget http://files.freeswitch.org/downloads/libs/json-c-0.9.tar.gz
--2010-05-12 11:34:00--  http://files.freeswitch.org/downloads/libs/json-c-0.9.tar.gz
Resolving files.freeswitch.org... 69.174.57.101
Connecting to files.freeswitch.org|69.174.57.101|:80... connected.
HTTP request sent, awaiting response... 200 OK
Length: 328935 (321K) [application/x-gzip]
Saving to: `json-c-0.9.tar.gz'

100%[======================================================================================================================>] 328,935      930K/s   in 0.3s    

2010-05-12 11:34:00 (930 KB/s) - `json-c-0.9.tar.gz' saved [328935/328935]



/b

On May 12, 2010, at 11:27 AM, Shrouk Khan wrote:

> making install mod_curl
> --2010-05-12 23:13:27--  http://files.freeswitch.org/downloads/libs/json-c-0.9.tar.gz
> Resolving files.freeswitch.org... 69.174.57.101
> Connecting to files.freeswitch.org|69.174.57.101|:80... failed: Connection refused.
> cannot find json-c-0.9.tar.gz
> make[5]: *** [/usr/src/freeswitch-1.0.6/libs/json-c-0.9] Error 1
> make[4]: *** [install] Error 1
> make[3]: *** [mod_curl-install] Error 1
> make[2]: *** [install-recursive] Error 1
> 

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From msc at freeswitch.org  Wed May 12 09:39:35 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 12 May 2010 09:39:35 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call Starting Shortly!
Message-ID: <AANLkTinNc1gf7lAzg2cfzoMN0ufyImaYiFQDMSOANu0q@mail.gmail.com>

C'mon down!

http://wiki.freeswitch.org/wiki/FS_weekly_2010_05_12

DRK is discussing mod_managed and we have a few other things to talk about.
Please bring your questions and your handy tips & tricks!

-MC
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From helmut.kuper at ewetel.de  Wed May 12 09:47:56 2010
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Wed, 12 May 2010 18:47:56 +0200
Subject: [Freeswitch-users] mod_openzap, sangoma, FAX and HW-HEC
Message-ID: <4BEADBBC.5090802@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hello,


I'm not sure whether I'm right here or not. But maybe someone has faced
same problem:


Scenario A:
I'm using Sangoma A104d and FreeSWITCH Version 1.0.head (17097:17188M).
I enabled hardware based detection of fax and dtmf. Further hardware
based HEC is enabled

I'm not able to reveive a FAX (analog) via PSTN, while sending works and
internal (SIP) works as well. So there is no problem with the ATAs I use
nor with the network (which routes the RTP along 80 kilometers one way).

Scenario B:
I'm able to receive FAX from pstn with same setup, but have hardware
based HEC disabled. So in this case everything works fine.


I thought sangoma's hardware based Fax detection feature should have
disabled HEC when a Fax is detected, but it doesn't seems so. So my
current thoughts are about the question whether FS is able to disable
hardware HEC via mod_openzap and wanpipe driver for the corresponding
A104d channel, when FS knows that the target is a FAX device ...


-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.7 (MingW32)

iD8DBQFL6tu84tZeNddg3dwRAur3AJ4rOw5ocXm+aGG7spLYtA4yx23eHACfcUPF
K+VHsbNiCR0XQeztzeI4iTY=
=ihsg
-----END PGP SIGNATURE-----


From m.krivushin at imarto.net  Wed May 12 10:21:42 2010
From: m.krivushin at imarto.net (Mikhail Krivushin)
Date: Thu, 13 May 2010 00:21:42 +0700
Subject: [Freeswitch-users] echo video calls
Message-ID: <k2u5be734a51005121021p81bee4a7o6df5ebb6b18696a2@mail.gmail.com>

Hello! I'm trying to build echo service with video support, but fail.

I add video codecs to `global_codec_prefs` and `outbound_codec_prefs`. Add
extension:
    <extension name="echo123">
      <condition field="destination_number" expression="^echo123$">
        <action application="answer"/>
        <action application="echo"/>
      </condition>
    </extension>
And try to make video call. My phone send SDP:

   v=0
   o=deepwalker 123456 654321 IN IP4
   s=A conversation
   c=IN IP4
   t=0 0
   m=audio 7078 RTP/AVP 111 110 112 3 0 8 101
   b=AS:80
   a=rtpmap:111 speex/16000/1
   a=rtpmap:110 speex/8000/1
   a=rtpmap:112 speex/32000/1
   a=rtpmap:3 GSM/8000/1
   a=rtpmap:0 PCMU/8000/1
   a=rtpmap:8 PCMA/8000/1
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-11
   m=video 9078 RTP/AVP 102 103 34 98 99 97
   b=AS:410
   a=rtpmap:102 H264/90000
   a=rtpmap:103 H264/90000
   a=fmtp:103 packetization-mode=1
   a=rtpmap:34 H263/90000
   a=rtpmap:98 H263-1998/90000
   a=fmtp:98 CIF=1;QCIF=1
   a=rtpmap:99 MP4V-ES/90000
   a=fmtp:99 profile-level-id=3
   a=rtpmap:97 theora/90000


But FreeSWITCH send back:

   v=0
   o=FreeSWITCH 1273664404 1273664405 IN IP4
   s=FreeSWITCH
   c=IN IP4
   t=0 0
   m=audio 20146 RTP/AVP 0 101
   a=rtpmap:0 PCMU/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=silenceSupp:off - - - -
   a=ptime:20
   m=video 0 RTP/AVP 19

Can anyone suggest me, what I need to do?

-- 
? ?????????, ???????? ??????
?. ????? ???. +7 913 865 78 66
icq: 218 744 127
xmpp: KrivushinME at jabber.ru
skype: mkrivushin
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From anthony.minessale at gmail.com  Wed May 12 10:24:54 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 12 May 2010 12:24:54 -0500
Subject: [Freeswitch-users] mod_openzap, sangoma, FAX and HW-HEC
In-Reply-To: <4BEADBBC.5090802@ewetel.de>
References: <4BEADBBC.5090802@ewetel.de>
Message-ID: <AANLkTiksacBjB7uuCzZidLX5PxVFmNsE7FHNskSVjhnM@mail.gmail.com>

openzap has an app you can run to disable EC but i don't know if it's
implemented into the sangoma plugin but its not hard to fix if it's not.


On Wed, May 12, 2010 at 11:47 AM, Helmut Kuper <helmut.kuper at ewetel.de>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hello,
>
>
> I'm not sure whether I'm right here or not. But maybe someone has faced
> same problem:
>
>
> Scenario A:
> I'm using Sangoma A104d and FreeSWITCH Version 1.0.head (17097:17188M).
> I enabled hardware based detection of fax and dtmf. Further hardware
> based HEC is enabled
>
> I'm not able to reveive a FAX (analog) via PSTN, while sending works and
> internal (SIP) works as well. So there is no problem with the ATAs I use
> nor with the network (which routes the RTP along 80 kilometers one way).
>
> Scenario B:
> I'm able to receive FAX from pstn with same setup, but have hardware
> based HEC disabled. So in this case everything works fine.
>
>
> I thought sangoma's hardware based Fax detection feature should have
> disabled HEC when a Fax is detected, but it doesn't seems so. So my
> current thoughts are about the question whether FS is able to disable
> hardware HEC via mod_openzap and wanpipe driver for the corresponding
> A104d channel, when FS knows that the target is a FAX device ...
>
>
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.7 (MingW32)
>
> iD8DBQFL6tu84tZeNddg3dwRAur3AJ4rOw5ocXm+aGG7spLYtA4yx23eHACfcUPF
> K+VHsbNiCR0XQeztzeI4iTY=
> =ihsg
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <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
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From brian at freeswitch.org  Wed May 12 10:26:02 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 12 May 2010 12:26:02 -0500
Subject: [Freeswitch-users] echo video calls
In-Reply-To: <k2u5be734a51005121021p81bee4a7o6df5ebb6b18696a2@mail.gmail.com>
References: <k2u5be734a51005121021p81bee4a7o6df5ebb6b18696a2@mail.gmail.com>
Message-ID: <EEC686D5-8839-49DF-82E4-57F613DDF2F6@freeswitch.org>

This means it rejected the video... did you allow any video codecs?

/b

On May 12, 2010, at 12:21 PM, Mikhail Krivushin wrote:

>    m=video 0 RTP/AVP 19



From lloyd.aloysius at gmail.com  Wed May 12 10:28:19 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Wed, 12 May 2010 13:28:19 -0400
Subject: [Freeswitch-users] Call Forwarding
Message-ID: <AANLkTilGDHjAG7moKpERFGdD8lm-V2vjlbzLvpk0HQ4-@mail.gmail.com>

Hi All,

How to Call Forward a Extension to another Extension or to a PSTN Number  in
FreeSWITCH?



Thanks
Lloyd
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From m.krivushin at imarto.net  Wed May 12 10:57:03 2010
From: m.krivushin at imarto.net (Mikhail Krivushin)
Date: Thu, 13 May 2010 00:57:03 +0700
Subject: [Freeswitch-users] echo video calls
In-Reply-To: <EEC686D5-8839-49DF-82E4-57F613DDF2F6@freeswitch.org>
References: <k2u5be734a51005121021p81bee4a7o6df5ebb6b18696a2@mail.gmail.com>
	<EEC686D5-8839-49DF-82E4-57F613DDF2F6@freeswitch.org>
Message-ID: <k2h5be734a51005121057xc91dd5b6u1fb992a91db7b783@mail.gmail.com>

Sorry, I dont install mod_h26x.

I use our new debian packaging -
http://github.com/vladimirelizarov/debian-freeswitch
This variant need more work to be shine, but it more debianized, then
default variant.
It cool, but dont include mod_h26x by default : )

Thank you for your attention, Brian.


2010/5/13 Brian West <brian at freeswitch.org>

> This means it rejected the video... did you allow any video codecs?
>
>
-- 
? ?????????, ???????? ??????
?. ????? ???. +7 913 865 78 66
icq: 218 744 127
xmpp: KrivushinME at jabber.ru
skype: mkrivushin
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From david.ponzone at gmail.com  Wed May 12 11:15:19 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Wed, 12 May 2010 20:15:19 +0200
Subject: [Freeswitch-users] Call Forwarding
In-Reply-To: <AANLkTilGDHjAG7moKpERFGdD8lm-V2vjlbzLvpk0HQ4-@mail.gmail.com>
References: <AANLkTilGDHjAG7moKpERFGdD8lm-V2vjlbzLvpk0HQ4-@mail.gmail.com>
Message-ID: <1183C5D6-D403-477C-A6EA-41526C160CA2@gmail.com>

Well, add the right rules to the dialplan.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 12/05/2010 ? 19:28, Aloysius Lloyd a ?crit :

> Hi All,
>
> How to Call Forward a Extension to another Extension or to a PSTN  
> Number  in FreeSWITCH?
>
>
>
> Thanks
> Lloyd
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jor3l at foravatars.com  Wed May 12 11:58:05 2010
From: jor3l at foravatars.com (Jor3l Boa)
Date: Wed, 12 May 2010 13:58:05 -0500
Subject: [Freeswitch-users] mod_sofia help
Message-ID: <AANLkTik-EfRDjgn8KpMENbkRD1HdMCmtxHBkuyA1AOXe@mail.gmail.com>

Hello there, I need some help with Sofia and FreeSwitch (new here :)) when
the client attempts to connect I'm getting this:

2010-05-12 13:51:16.888628 [WARNING] sofia_reg.c:1031 SIP auth failure
(REGISTER) on sofia profile 'internal' for [xbd6Y0PU7RI2mIOBsPW83Gw==@
192.168.0.171] from ip 192.168.0.158
2010-05-12 13:51:39.707814 [WARNING] sofia_reg.c:1879 Can't find user
[xbd6Y0PU7RI2mIOBsPW83Gw==@192.168.0.171]
You must define a domain called '192.168.0.171' in your directory and add a
user with the id="xbd6Y0PU7RI2mIOBsPW83Gw==" attribute
and you must configure your device to use the proper domain in it's
authentication credentials.
2010-05-12 13:51:39.710879 [WARNING] sofia_reg.c:1031 SIP auth failure
(REGISTER) on sofia profile 'internal' for [xbd6Y0PU7RI2mIOBsPW83Gw==@
192.168.0.171] from ip 192.168.0.158

I guess there is a command to allow any user using this port but I'm not
sure.. tried google with no luck.

So, is there a way to allow any user to use the service or be registered
automatically? since each user name, is different and I can't add each one
manually (it's a lot of info), or, is possible to shutdown Sofia and use the
service without it?

Thanks  :-)
Jorel
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From neilp at cs.stanford.edu  Wed May 12 11:59:26 2010
From: neilp at cs.stanford.edu (Neil Patel)
Date: Wed, 12 May 2010 11:59:26 -0700
Subject: [Freeswitch-users] how do I detect a call has rung once?
In-Reply-To: <AANLkTilcSOp1nfZxSqtndA_HBvKNl1iPrOwJk4uC3P4z@mail.gmail.com>
References: <u2je5503bdb1005101403u405a5eebo9a2b8c698e9cffaf@mail.gmail.com> 
	<AANLkTinS6No0-1Z1oe9qF6T_9yVedPl3JI-T2OopEoep@mail.gmail.com> 
	<201005101734.09903.sos@sokhapkin.dyndns.org>
	<AANLkTilVHAdZDgF3a5Mg2HgXrXOg6hkrLKGb9xUAAdik@mail.gmail.com> 
	<AANLkTikqOpUTWkv9izTLM8CKljgexflNPPEVUxiYzw01@mail.gmail.com> 
	<E08F3AA9-613E-446A-BD22-5B31F38EB78C@freeswitch.org>
	<AANLkTimHH6zzA_-Dal1YkHlSL4gkbp7m8TyoFrukBmB3@mail.gmail.com> 
	<AANLkTinHQnCTPTDtxXRl80Pd9aGMs8V2Xws1z5KeMegU@mail.gmail.com> 
	<AANLkTilcSOp1nfZxSqtndA_HBvKNl1iPrOwJk4uC3P4z@mail.gmail.com>
Message-ID: <AANLkTiklAqw6Er_0fCq9CQnt6RyEPY9P0Usg5iAjl4nw@mail.gmail.com>

Thanks for the tips.

Where are there examples of using ESL to register and handle events? For
e.g. I don't see any example of how to register for a change in channel
variable endpoint_disposition. Or even to check the CHANNEL_STATE event.

-Neil

On Tue, May 11, 2010 at 8:35 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

>
> you don't poll for it you wait for the events.
> polling variables is horribly inefficient
>
> On Mon, May 10, 2010 at 10:11 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> doh
>> I mean ${endpoint_disposition}
>>
>>
>> On Mon, May 10, 2010 at 8:10 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> Off the top of my head possibly it's ${endpoint_dispostion}
>>> It's been quite a while since I did my openzap stuff...
>>> -MC
>>>
>>>
>>> On Mon, May 10, 2010 at 6:48 PM, Brian West <brian at freeswitch.org>wrote:
>>>
>>>> yes you will get progressing and alerting but I no not recall which
>>>> those map to.
>>>>
>>>> /b
>>>>
>>>>
>>>> Sent from my iPad
>>>>
>>>> On May 10, 2010, at 8:35 PM, Neil Patel <neilp at cs.stanford.edu> wrote:
>>>>
>>>> > What's the channel variable I monitor for? Is it good practice to poll
>>>> for it, or is there another option? Also, does 180 and/or 183 come back for
>>>> non-sip calls (I'm dialing out over PRI/openzap).
>>>> >
>>>> > Thanks,
>>>> > Neil
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Wed May 12 12:11:20 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 12 May 2010 14:11:20 -0500
Subject: [Freeswitch-users] mod_sofia help
In-Reply-To: <AANLkTik-EfRDjgn8KpMENbkRD1HdMCmtxHBkuyA1AOXe@mail.gmail.com>
References: <AANLkTik-EfRDjgn8KpMENbkRD1HdMCmtxHBkuyA1AOXe@mail.gmail.com>
Message-ID: <AANLkTinfJHnm3bbEaovUJCp34z09uBsF3cPZ6bjWu2fT@mail.gmail.com>

set the blind-reg param which is commented out in the default config
unset the auth-calls

then you can just blindly reg any ext you want and they can call each other

On Wed, May 12, 2010 at 1:58 PM, Jor3l Boa <jor3l at foravatars.com> wrote:

> Hello there, I need some help with Sofia and FreeSwitch (new here :)) when
> the client attempts to connect I'm getting this:
>
> 2010-05-12 13:51:16.888628 [WARNING] sofia_reg.c:1031 SIP auth failure
> (REGISTER) on sofia profile 'internal' for [xbd6Y0PU7RI2mIOBsPW83Gw==@
> 192.168.0.171] from ip 192.168.0.158
> 2010-05-12 13:51:39.707814 [WARNING] sofia_reg.c:1879 Can't find user
> [xbd6Y0PU7RI2mIOBsPW83Gw==@192.168.0.171]
> You must define a domain called '192.168.0.171' in your directory and add a
> user with the id="xbd6Y0PU7RI2mIOBsPW83Gw==" attribute
> and you must configure your device to use the proper domain in it's
> authentication credentials.
> 2010-05-12 13:51:39.710879 [WARNING] sofia_reg.c:1031 SIP auth failure
> (REGISTER) on sofia profile 'internal' for [xbd6Y0PU7RI2mIOBsPW83Gw==@
> 192.168.0.171] from ip 192.168.0.158
>
> I guess there is a command to allow any user using this port but I'm not
> sure.. tried google with no luck.
>
> So, is there a way to allow any user to use the service or be registered
> automatically? since each user name, is different and I can't add each one
> manually (it's a lot of info), or, is possible to shutdown Sofia and use the
> service without it?
>
> Thanks  :-)
> Jorel
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <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
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From anatoliy at kounitskiy.com  Wed May 12 12:22:32 2010
From: anatoliy at kounitskiy.com (Anatoliy Kounitskiy)
Date: Wed, 12 May 2010 22:22:32 +0300
Subject: [Freeswitch-users] Call Forwarding
In-Reply-To: <AANLkTilGDHjAG7moKpERFGdD8lm-V2vjlbzLvpk0HQ4-@mail.gmail.com>
References: <AANLkTilGDHjAG7moKpERFGdD8lm-V2vjlbzLvpk0HQ4-@mail.gmail.com>
Message-ID: <1273692152.1907.6.camel@lenovo400>

Check some of the following ways:
Deflect - http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect
Transfer - http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_transfer
Redirect - http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect

And of course, make the appropriate contexts/logic in your dialplan.

Regards,
Anatoliy 

On Wed, 2010-05-12 at 13:28 -0400, Aloysius Lloyd wrote:
> Hi All, 
> 
> 
> How to Call Forward a Extension to another Extension or to a PSTN
> Number  in FreeSWITCH?
> 
> 
> 
> 
> 
> 
> Thanks
> Lloyd
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Anatoliy Kounitskiy
-------------------------
E-mail: anatoliy at kounitskiy.com
Mobile: +359898913540



From lloyd.aloysius at sunteltech.ca  Wed May 12 12:35:02 2010
From: lloyd.aloysius at sunteltech.ca (Aloysius Lloyd)
Date: Wed, 12 May 2010 15:35:02 -0400
Subject: [Freeswitch-users] Call Forwarding
In-Reply-To: <1273692152.1907.6.camel@lenovo400>
References: <AANLkTilGDHjAG7moKpERFGdD8lm-V2vjlbzLvpk0HQ4-@mail.gmail.com> 
	<1273692152.1907.6.camel@lenovo400>
Message-ID: <AANLkTimuU8hCTI3OiruByOZwhvNVgnoA4Abf66Xrx66h@mail.gmail.com>

Thank you for the suggestions. But I am looking for a way to store the
call forward number.

What is the recommended place to store the call forwarded number.

*72 Activate call forward

Ask for the number then save it to the database then use the number for call
forwarded.

*73  Deactivate call forward

Delete the call forward number from the database.

Thanks
Lloyd


On Wed, May 12, 2010 at 3:22 PM, Anatoliy Kounitskiy <
anatoliy at kounitskiy.com> wrote:

> Check some of the following ways:
> Deflect - http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect
> Transfer - http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_transfer
> Redirect - http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect
>
> And of course, make the appropriate contexts/logic in your dialplan.
>
> Regards,
> Anatoliy
>
> On Wed, 2010-05-12 at 13:28 -0400, Aloysius Lloyd wrote:
> > Hi All,
> >
> >
> > How to Call Forward a Extension to another Extension or to a PSTN
> > Number  in FreeSWITCH?
> >
> >
> >
> >
> >
> >
> > Thanks
> > Lloyd
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> --
> Anatoliy Kounitskiy
> -------------------------
> E-mail: anatoliy at kounitskiy.com
> Mobile: +359898913540
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From moises.silva at gmail.com  Wed May 12 12:40:19 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Wed, 12 May 2010 15:40:19 -0400
Subject: [Freeswitch-users] mod_openzap, sangoma, FAX and HW-HEC
In-Reply-To: <AANLkTiksacBjB7uuCzZidLX5PxVFmNsE7FHNskSVjhnM@mail.gmail.com>
References: <4BEADBBC.5090802@ewetel.de>
	<AANLkTiksacBjB7uuCzZidLX5PxVFmNsE7FHNskSVjhnM@mail.gmail.com>
Message-ID: <AANLkTimVaDRo7ypzZu4YmgeVVMdV-sQ4TEocJOHaFv-F@mail.gmail.com>

It is implemented and should work as a work-around.

Having said that, I suggest you to contact Sangoma support (
techdesk at sangoma.com), the ec chip should detect fax tone and disable its
echo-canceling functions. However, some old versions of the driver have a
noise-reduction feature enabled by default which also happen to disrupt
faxes.

Try adding this just below TDMV_HWEC=YES option:

HWEC_NOISE_REDUCTION_DISABLE=YES

Or upgrade to the latest driver which does this automatically.

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com


On Wed, May 12, 2010 at 1:24 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> openzap has an app you can run to disable EC but i don't know if it's
> implemented into the sangoma plugin but its not hard to fix if it's not.
>
>
> On Wed, May 12, 2010 at 11:47 AM, Helmut Kuper <helmut.kuper at ewetel.de>wrote:
>
>> -----BEGIN PGP SIGNED MESSAGE-----
>> Hash: SHA1
>>
>> Hello,
>>
>>
>> I'm not sure whether I'm right here or not. But maybe someone has faced
>> same problem:
>>
>>
>> Scenario A:
>> I'm using Sangoma A104d and FreeSWITCH Version 1.0.head (17097:17188M).
>> I enabled hardware based detection of fax and dtmf. Further hardware
>> based HEC is enabled
>>
>> I'm not able to reveive a FAX (analog) via PSTN, while sending works and
>> internal (SIP) works as well. So there is no problem with the ATAs I use
>> nor with the network (which routes the RTP along 80 kilometers one way).
>>
>> Scenario B:
>> I'm able to receive FAX from pstn with same setup, but have hardware
>> based HEC disabled. So in this case everything works fine.
>>
>>
>> I thought sangoma's hardware based Fax detection feature should have
>> disabled HEC when a Fax is detected, but it doesn't seems so. So my
>> current thoughts are about the question whether FS is able to disable
>> hardware HEC via mod_openzap and wanpipe driver for the corresponding
>> A104d channel, when FS knows that the target is a FAX device ...
>>
>>
>> -----BEGIN PGP SIGNATURE-----
>> Version: GnuPG v1.4.7 (MingW32)
>>
>> iD8DBQFL6tu84tZeNddg3dwRAur3AJ4rOw5ocXm+aGG7spLYtA4yx23eHACfcUPF
>> K+VHsbNiCR0XQeztzeI4iTY=
>> =ihsg
>> -----END PGP SIGNATURE-----
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From moises.silva at gmail.com  Wed May 12 12:47:37 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Wed, 12 May 2010 15:47:37 -0400
Subject: [Freeswitch-users] mod_openzap, sangoma, FAX and HW-HEC
In-Reply-To: <AANLkTimVaDRo7ypzZu4YmgeVVMdV-sQ4TEocJOHaFv-F@mail.gmail.com>
References: <4BEADBBC.5090802@ewetel.de>
	<AANLkTiksacBjB7uuCzZidLX5PxVFmNsE7FHNskSVjhnM@mail.gmail.com>
	<AANLkTimVaDRo7ypzZu4YmgeVVMdV-sQ4TEocJOHaFv-F@mail.gmail.com>
Message-ID: <AANLkTikUPqLyeeToYSj9SeJEdkz5cLsMT6_r6ZjEJyc3@mail.gmail.com>

Sorry, the setting should be under [wanpipex] in wanpipex.conf file and NOT
in the [wxgx] section.

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com


On Wed, May 12, 2010 at 3:40 PM, Moises Silva <moises.silva at gmail.com>wrote:

> It is implemented and should work as a work-around.
>
> Having said that, I suggest you to contact Sangoma support (
> techdesk at sangoma.com), the ec chip should detect fax tone and disable its
> echo-canceling functions. However, some old versions of the driver have a
> noise-reduction feature enabled by default which also happen to disrupt
> faxes.
>
> Try adding this just below TDMV_HWEC=YES option:
>
> HWEC_NOISE_REDUCTION_DISABLE=YES
>
> Or upgrade to the latest driver which does this automatically.
>
> Moises Silva
> Senior Software Engineer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
> 9T3 Canada
> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>
>
>
> On Wed, May 12, 2010 at 1:24 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> openzap has an app you can run to disable EC but i don't know if it's
>> implemented into the sangoma plugin but its not hard to fix if it's not.
>>
>>
>> On Wed, May 12, 2010 at 11:47 AM, Helmut Kuper <helmut.kuper at ewetel.de>wrote:
>>
>>> -----BEGIN PGP SIGNED MESSAGE-----
>>> Hash: SHA1
>>>
>>> Hello,
>>>
>>>
>>> I'm not sure whether I'm right here or not. But maybe someone has faced
>>> same problem:
>>>
>>>
>>> Scenario A:
>>> I'm using Sangoma A104d and FreeSWITCH Version 1.0.head (17097:17188M).
>>> I enabled hardware based detection of fax and dtmf. Further hardware
>>> based HEC is enabled
>>>
>>> I'm not able to reveive a FAX (analog) via PSTN, while sending works and
>>> internal (SIP) works as well. So there is no problem with the ATAs I use
>>> nor with the network (which routes the RTP along 80 kilometers one way).
>>>
>>> Scenario B:
>>> I'm able to receive FAX from pstn with same setup, but have hardware
>>> based HEC disabled. So in this case everything works fine.
>>>
>>>
>>> I thought sangoma's hardware based Fax detection feature should have
>>> disabled HEC when a Fax is detected, but it doesn't seems so. So my
>>> current thoughts are about the question whether FS is able to disable
>>> hardware HEC via mod_openzap and wanpipe driver for the corresponding
>>> A104d channel, when FS knows that the target is a FAX device ...
>>>
>>>
>>> -----BEGIN PGP SIGNATURE-----
>>> Version: GnuPG v1.4.7 (MingW32)
>>>
>>> iD8DBQFL6tu84tZeNddg3dwRAur3AJ4rOw5ocXm+aGG7spLYtA4yx23eHACfcUPF
>>> K+VHsbNiCR0XQeztzeI4iTY=
>>> =ihsg
>>> -----END PGP SIGNATURE-----
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <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-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From anatoliy at kounitskiy.com  Wed May 12 14:01:47 2010
From: anatoliy at kounitskiy.com (Anatoliy Kounitskiy)
Date: Thu, 13 May 2010 00:01:47 +0300
Subject: [Freeswitch-users] Call Forwarding
In-Reply-To: <AANLkTimuU8hCTI3OiruByOZwhvNVgnoA4Abf66Xrx66h@mail.gmail.com>
References: <AANLkTilGDHjAG7moKpERFGdD8lm-V2vjlbzLvpk0HQ4-@mail.gmail.com>
	<1273692152.1907.6.camel@lenovo400>
	<AANLkTimuU8hCTI3OiruByOZwhvNVgnoA4Abf66Xrx66h@mail.gmail.com>
Message-ID: <1273698107.1907.10.camel@lenovo400>

:) Probably, you've already found mod_limit
http://wiki.freeswitch.org/wiki/Mod_limit and the db function.

I searched also for the same thing couple of weeks ago. And I decided to
go with the old "asterisk" way - use the db for this.

Probably there is a nicer way to do this, but ... still searching.

Regards,
Anatoliy


On Wed, 2010-05-12 at 15:35 -0400, Aloysius Lloyd wrote:
> Thank you for the suggestions. But I am looking for a way to store the
> call forward number.
> 
> 
> What is the recommended place to store the call forwarded number.
> 
> 
> *72 Activate call forward
> 
> 
> Ask for the number then save it to the database then use the number
> for call forwarded.
> 
> 
> *73  Deactivate call forward
> 
> 
> Delete the call forward number from the database.
> 
> 
> Thanks
> Lloyd
> 
> 
> On Wed, May 12, 2010 at 3:22 PM, Anatoliy Kounitskiy
> <anatoliy at kounitskiy.com> wrote:
>         Check some of the following ways:
>         Deflect -
>         http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect
>         Transfer -
>         http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_transfer
>         Redirect -
>         http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect
>         
>         And of course, make the appropriate contexts/logic in your
>         dialplan.
>         
>         Regards,
>         Anatoliy
>         
>         
>         On Wed, 2010-05-12 at 13:28 -0400, Aloysius Lloyd wrote:
>         > Hi All,
>         >
>         >
>         > How to Call Forward a Extension to another Extension or to a
>         PSTN
>         > Number  in FreeSWITCH?
>         >
>         >
>         >
>         >
>         >
>         >
>         > Thanks
>         > Lloyd
>         
>         > _______________________________________________
>         > FreeSWITCH-users mailing list
>         > FreeSWITCH-users at lists.freeswitch.org
>         >
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         >
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         > http://www.freeswitch.org
>         
>         
>         --
>         Anatoliy Kounitskiy
>         -------------------------
>         E-mail: anatoliy at kounitskiy.com
>         Mobile: +359898913540
>         
>         
>         
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>         
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Anatoliy Kounitskiy
-------------------------
E-mail: anatoliy at kounitskiy.com
Mobile: +359898913540



From jeff at jefflenk.com  Wed May 12 14:32:38 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 12 May 2010 14:32:38 -0700 (PDT)
Subject: [Freeswitch-users] build error on latest git
In-Reply-To: <AANLkTikcvi3S-mW7IJR_bU4v9gwmdTwECmRypMjX-G3J@mail.gmail.com>
References: <AANLkTikcvi3S-mW7IJR_bU4v9gwmdTwECmRypMjX-G3J@mail.gmail.com>
Message-ID: <1273699958517-5043496.post@n2.nabble.com>


This was fixed today - git head
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/build-error-on-latest-git-tp5041094p5043496.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From phone.bytes at gmail.com  Wed May 12 12:40:58 2010
From: phone.bytes at gmail.com (Phone)
Date: Wed, 12 May 2010 13:40:58 -0600
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
Message-ID: <4BEB044A.6030906@gmail.com>

I am looking for a boot in the right direction on the following project.

I am looking to build an "application" with FS where it will frequently
check a database and then make an outbound call when a record with
certain conditions is found in the DB.  It needs to determine if the
call was then answered by a machine/vm, busy, or answered by a person,
play a message and get a response back to be recorded in the db.  It may
find many calls that need to be made at the same time.  It will be
running on a PRI.

I am confused as to what approach to use to accomplish this.

Lua Script, Perl Script, Javascript, Phython, PHP, Ruby....etc.
Liverpie, ESL....etc.
Or a totally different approach.

I realize that there is not only one "correct" way to build this, but I
am looking for advise from those who have done this type of thing
before.  I value the opinion of those who have gone down this road and
would be willing to share their thoughts on a recommended path to take
to accomplish this.

Thanks



From mike at jerris.com  Wed May 12 14:39:30 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 12 May 2010 17:39:30 -0400
Subject: [Freeswitch-users] echo video calls
In-Reply-To: <k2h5be734a51005121057xc91dd5b6u1fb992a91db7b783@mail.gmail.com>
References: <k2u5be734a51005121021p81bee4a7o6df5ebb6b18696a2@mail.gmail.com>
	<EEC686D5-8839-49DF-82E4-57F613DDF2F6@freeswitch.org>
	<k2h5be734a51005121057xc91dd5b6u1fb992a91db7b783@mail.gmail.com>
Message-ID: <F7BF1196-D29A-4508-B81F-7709C288BAD7@jerris.com>

Would you mind putting together patches that we can merge into tree so you don't have to hold out of tree patches.  For the configure patches, we should do something like we did with sysconfig dir where it uses the standard autoconf switches, but remains with the current defaults.  As for the downloading patch, we should make it use the tarballs if they are there already, which should obsolete this patch.  As for the rest of the changes, I would rather have as much of this merged into tree, but the ones in tree should use our default directory layout.  This should leave you with a minor patch or better just a single option to change the directory layout.

Mike


On May 12, 2010, at 1:57 PM, Mikhail Krivushin wrote:

> Sorry, I dont install mod_h26x.
> 
> I use our new debian packaging - http://github.com/vladimirelizarov/debian-freeswitch
> This variant need more work to be shine, but it more debianized, then default variant.
> It cool, but dont include mod_h26x by default : ) 
> 
> Thank you for your attention, Brian.
> 
> 
> 2010/5/13 Brian West <brian at freeswitch.org>
> This means it rejected the video... did you allow any video codecs?
> 
> 
> -- 
> ? ?????????, ???????? ??????
> ?. ????? ???. +7 913 865 78 66
> icq: 218 744 127
> xmpp: KrivushinME at jabber.ru
> skype: mkrivushin
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From kris at kriskinc.com  Wed May 12 14:50:17 2010
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 12 May 2010 17:50:17 -0400
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
Message-ID: <0b27b9ae92e848c6edfcb0b6b20d255f@mail.gmail.com>

Are you going to be calling my cell phone to ask if I owe $10,000 or more to
the IRS?

Seriously, the world does not need another robodialer to harass people...

With that being said, write your app in whatever you want using ESL and
connect to the socket to originate calls and listen for events.


--
Kristian Kielhofner
http://blog.krisk.org

----- Original Message -----
From: freeswitch-users-bounces at lists.freeswitch.org
<freeswitch-users-bounces at lists.freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
<freeswitch-users at lists.freeswitch.org>
Sent: Wed May 12 15:40:58 2010
Subject: [Freeswitch-users] Questions on Building an application
forFreeSWITCHI am looking for a boot in the right direction on the
following project.

I am looking to build an "application" with FS where it will frequently
check a database and then make an outbound call when a record with
certain conditions is found in the DB.  It needs to determine if the
call was then answered by a machine/vm, busy, or answered by a person,
play a message and get a response back to be recorded in the db.  It may
find many calls that need to be made at the same time.  It will be
running on a PRI.

I am confused as to what approach to use to accomplish this.

Lua Script, Perl Script, Javascript, Phython, PHP, Ruby....etc.
Liverpie, ESL....etc.
Or a totally different approach.

I realize that there is not only one "correct" way to build this, but I
am looking for advise from those who have done this type of thing
before.  I value the opinion of those who have gone down this road and
would be willing to share their thoughts on a recommended path to take
to accomplish this.

Thanks


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


From msc at freeswitch.org  Wed May 12 14:47:41 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 12 May 2010 14:47:41 -0700
Subject: [Freeswitch-users] how do I detect a call has rung once?
In-Reply-To: <AANLkTiklAqw6Er_0fCq9CQnt6RyEPY9P0Usg5iAjl4nw@mail.gmail.com>
References: <u2je5503bdb1005101403u405a5eebo9a2b8c698e9cffaf@mail.gmail.com>
	<AANLkTinS6No0-1Z1oe9qF6T_9yVedPl3JI-T2OopEoep@mail.gmail.com>
	<201005101734.09903.sos@sokhapkin.dyndns.org>
	<AANLkTilVHAdZDgF3a5Mg2HgXrXOg6hkrLKGb9xUAAdik@mail.gmail.com>
	<AANLkTikqOpUTWkv9izTLM8CKljgexflNPPEVUxiYzw01@mail.gmail.com>
	<E08F3AA9-613E-446A-BD22-5B31F38EB78C@freeswitch.org>
	<AANLkTimHH6zzA_-Dal1YkHlSL4gkbp7m8TyoFrukBmB3@mail.gmail.com>
	<AANLkTinHQnCTPTDtxXRl80Pd9aGMs8V2Xws1z5KeMegU@mail.gmail.com>
	<AANLkTilcSOp1nfZxSqtndA_HBvKNl1iPrOwJk4uC3P4z@mail.gmail.com>
	<AANLkTiklAqw6Er_0fCq9CQnt6RyEPY9P0Usg5iAjl4nw@mail.gmail.com>
Message-ID: <AANLkTinnTRKaOUI2OHSoi9Lu1jjbSfTxZjFBn_9EIMa1@mail.gmail.com>

On Wed, May 12, 2010 at 11:59 AM, Neil Patel <neilp at cs.stanford.edu> wrote:

> Thanks for the tips.
>
> Where are there examples of using ESL to register and handle events? For
> e.g. I don't see any example of how to register for a change in channel
> variable endpoint_disposition. Or even to check the CHANNEL_STATE event.
>
> -Neil
>
> Neil,

Check out the event socket documentation, specifically the "events plain
all" and "filter" commands. You can listen to all events or just specific
ones:
http://wiki.freeswitch.org/wiki/Event_socket

If you want to experiment with the event socket and the various events then
use fs_cli. Connect with fs_cli and then issue "/log 0" which turns off all
debug messages. Now you have basically a raw event socket. Try some of these
commands:

/event plain all

watch the fun messages fly. :)

You can filter them as well:

/filter Event-Name CHANNEL_STATE
/filter Event-Name CHANNEL_HANGUP

when you apply a filter you will receive only those things you choose. A
it's a "filter in" not filter out. Another way of saying it is "show me
events named CHANNEL_STATE" and "show me events named CHANNEL_HANGUP"

just note that in your script you will be using the ESL abstractions for
these:

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

I hope that helps! Have fun.
-MC
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From msc at freeswitch.org  Wed May 12 14:58:53 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 12 May 2010 14:58:53 -0700
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
In-Reply-To: <4BEB044A.6030906@gmail.com>
References: <4BEB044A.6030906@gmail.com>
Message-ID: <AANLkTimtCp-pT8IFLe3HEkaZ43HYzh9pP0gfD6sl5lbv@mail.gmail.com>

This brings back memories...

Okay, I actually did this sort of thing a few years back with a PRI and
proprietary db. It wasn't polished but it did work (mostly). Let me just say
that the FreeSWITCH side of things was like 10% of the project. The real
work is done in the "controller" application, the database, and whatever
mechanism you use to update the database with new call campaigns (or single
calls).

Do you have programming resources at your disposal to throw at this project?
If not I recommend consulting at freeswitch.org. (You will need to contact them
anyway if you want the ans mach detection module - it is not free.)

-MC

On Wed, May 12, 2010 at 12:40 PM, Phone <phone.bytes at gmail.com> wrote:

> I am looking for a boot in the right direction on the following project.
>
> I am looking to build an "application" with FS where it will frequently
> check a database and then make an outbound call when a record with
> certain conditions is found in the DB.  It needs to determine if the
> call was then answered by a machine/vm, busy, or answered by a person,
> play a message and get a response back to be recorded in the db.  It may
> find many calls that need to be made at the same time.  It will be
> running on a PRI.
>
> I am confused as to what approach to use to accomplish this.
>
> Lua Script, Perl Script, Javascript, Phython, PHP, Ruby....etc.
> Liverpie, ESL....etc.
> Or a totally different approach.
>
> I realize that there is not only one "correct" way to build this, but I
> am looking for advise from those who have done this type of thing
> before.  I value the opinion of those who have gone down this road and
> would be willing to share their thoughts on a recommended path to take
> to accomplish this.
>
> Thanks
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Wed May 12 15:02:14 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 12 May 2010 15:02:14 -0700
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
In-Reply-To: <0b27b9ae92e848c6edfcb0b6b20d255f@mail.gmail.com>
References: <0b27b9ae92e848c6edfcb0b6b20d255f@mail.gmail.com>
Message-ID: <AANLkTiloS2j7QKkitjSmhECZXANvXnX8NH3fZiBL1qnm@mail.gmail.com>

On Wed, May 12, 2010 at 2:50 PM, Kristian Kielhofner <kris at kriskinc.com>wrote:

> Are you going to be calling my cell phone to ask if I owe $10,000 or more
> to
> the IRS?
>
> Seriously, the world does not need another robodialer to harass people...
>
Agreed, however there are legitimate use cases for this, like a retailer
calling to say an order is in, or reminder calls sent out by a doctor's
office. But yes, if it's dialing for dollars then I'm with you: Por favor!
No mas!
-MC


>
> With that being said, write your app in whatever you want using ESL and
> connect to the socket to originate calls and listen for events.
>
>
> --
> Kristian Kielhofner
> http://blog.krisk.org
>
> ----- Original Message -----
> From: freeswitch-users-bounces at lists.freeswitch.org
> <freeswitch-users-bounces at lists.freeswitch.org>
> To: freeswitch-users at lists.freeswitch.org
> <freeswitch-users at lists.freeswitch.org>
> Sent: Wed May 12 15:40:58 2010
> Subject: [Freeswitch-users] Questions on Building an application
> forFreeSWITCHI am looking for a boot in the right direction on the
> following project.
>
> I am looking to build an "application" with FS where it will frequently
> check a database and then make an outbound call when a record with
> certain conditions is found in the DB.  It needs to determine if the
> call was then answered by a machine/vm, busy, or answered by a person,
> play a message and get a response back to be recorded in the db.  It may
> find many calls that need to be made at the same time.  It will be
> running on a PRI.
>
> I am confused as to what approach to use to accomplish this.
>
> Lua Script, Perl Script, Javascript, Phython, PHP, Ruby....etc.
> Liverpie, ESL....etc.
> Or a totally different approach.
>
> I realize that there is not only one "correct" way to build this, but I
> am looking for advise from those who have done this type of thing
> before.  I value the opinion of those who have gone down this road and
> would be willing to share their thoughts on a recommended path to take
> to accomplish this.
>
> Thanks
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From phone.bytes at gmail.com  Wed May 12 15:28:13 2010
From: phone.bytes at gmail.com (Phone)
Date: Wed, 12 May 2010 16:28:13 -0600
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
In-Reply-To: <AANLkTiloS2j7QKkitjSmhECZXANvXnX8NH3fZiBL1qnm@mail.gmail.com>
References: <0b27b9ae92e848c6edfcb0b6b20d255f@mail.gmail.com>
	<AANLkTiloS2j7QKkitjSmhECZXANvXnX8NH3fZiBL1qnm@mail.gmail.com>
Message-ID: <4BEB2B7D.4080406@gmail.com>

This is legit.  We are not into harassing anyone.  There are too many 
people doing that already.

We currently have this working on a Windows/Dialogic platform, but would 
like to get it over to FreeSWITCH for a number of reasons.  It is always 
nice to avoid spending alot of time developing using a method that you 
later find has serious issues or limitations and then having to change 
horses and start over.  I just thought I would tap on someone else's 
wisdom in regards to a good way to talk to FS that works well.  I did 
notice that the wiki talks like LUA is the preferred way to go.

Thanks



Michael Collins wrote:
>
>
> On Wed, May 12, 2010 at 2:50 PM, Kristian Kielhofner 
> <kris at kriskinc.com <mailto:kris at kriskinc.com>> wrote:
>
>     Are you going to be calling my cell phone to ask if I owe $10,000
>     or more to
>     the IRS?
>
>     Seriously, the world does not need another robodialer to harass
>     people...
>
> Agreed, however there are legitimate use cases for this, like a 
> retailer calling to say an order is in, or reminder calls sent out by 
> a doctor's office. But yes, if it's dialing for dollars then I'm with 
> you: Por favor! No mas!
> -MC
>  
>
>
>     With that being said, write your app in whatever you want using
>     ESL and
>     connect to the socket to originate calls and listen for events.
>
>
>     --
>     Kristian Kielhofner
>     http://blog.krisk.org
>
>     ----- Original Message -----
>     From: freeswitch-users-bounces at lists.freeswitch.org
>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>
>     <freeswitch-users-bounces at lists.freeswitch.org
>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>>
>     To: freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>
>     <freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>>
>     Sent: Wed May 12 15:40:58 2010
>     Subject: [Freeswitch-users] Questions on Building an application
>     forFreeSWITCHI am looking for a boot in the right direction on the
>     following project.
>
>     I am looking to build an "application" with FS where it will
>     frequently
>     check a database and then make an outbound call when a record with
>     certain conditions is found in the DB.  It needs to determine if the
>     call was then answered by a machine/vm, busy, or answered by a person,
>     play a message and get a response back to be recorded in the db.
>      It may
>     find many calls that need to be made at the same time.  It will be
>     running on a PRI.
>
>     I am confused as to what approach to use to accomplish this.
>
>     Lua Script, Perl Script, Javascript, Phython, PHP, Ruby....etc.
>     Liverpie, ESL....etc.
>     Or a totally different approach.
>
>     I realize that there is not only one "correct" way to build this,
>     but I
>     am looking for advise from those who have done this type of thing
>     before.  I value the opinion of those who have gone down this road and
>     would be willing to share their thoughts on a recommended path to take
>     to accomplish this.
>
>     Thanks
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From kris at kriskinc.com  Wed May 12 15:41:44 2010
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 12 May 2010 18:41:44 -0400
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
Message-ID: <490aedfdd7c8ac680bbe0d7f68c786de@mail.gmail.com>

Glad to hear.

I still say to build the app and logic completely separately and connect it
to FreeSWITCH over the socket using ESL.


--
Kristian Kielhofner
http://blog.krisk.org

----- Original Message -----
From: freeswitch-users-bounces at lists.freeswitch.org
<freeswitch-users-bounces at lists.freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
<freeswitch-users at lists.freeswitch.org>
Sent: Wed May 12 18:28:13 2010
Subject: Re: [Freeswitch-users] Questions on Building an application
forFreeSWITCHThis is legit.  We are not into harassing anyone.  There
are too many
people doing that already.

We currently have this working on a Windows/Dialogic platform, but would
like to get it over to FreeSWITCH for a number of reasons.  It is always
nice to avoid spending alot of time developing using a method that you
later find has serious issues or limitations and then having to change
horses and start over.  I just thought I would tap on someone else's
wisdom in regards to a good way to talk to FS that works well.  I did
notice that the wiki talks like LUA is the preferred way to go.

Thanks



Michael Collins wrote:
>
>
> On Wed, May 12, 2010 at 2:50 PM, Kristian Kielhofner
> <kris at kriskinc.com <mailto:kris at kriskinc.com>> wrote:
>
>     Are you going to be calling my cell phone to ask if I owe $10,000
>     or more to
>     the IRS?
>
>     Seriously, the world does not need another robodialer to harass
>     people...
>
> Agreed, however there are legitimate use cases for this, like a
> retailer calling to say an order is in, or reminder calls sent out by
> a doctor's office. But yes, if it's dialing for dollars then I'm with
> you: Por favor! No mas!
> -MC
>
>
>
>     With that being said, write your app in whatever you want using
>     ESL and
>     connect to the socket to originate calls and listen for events.
>
>
>     --
>     Kristian Kielhofner
>     http://blog.krisk.org
>
>     ----- Original Message -----
>     From: freeswitch-users-bounces at lists.freeswitch.org
>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>
>     <freeswitch-users-bounces at lists.freeswitch.org
>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>>
>     To: freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>
>     <freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>>
>     Sent: Wed May 12 15:40:58 2010
>     Subject: [Freeswitch-users] Questions on Building an application
>     forFreeSWITCHI am looking for a boot in the right direction on the
>     following project.
>
>     I am looking to build an "application" with FS where it will
>     frequently
>     check a database and then make an outbound call when a record with
>     certain conditions is found in the DB.  It needs to determine if the
>     call was then answered by a machine/vm, busy, or answered by a person,
>     play a message and get a response back to be recorded in the db.
>      It may
>     find many calls that need to be made at the same time.  It will be
>     running on a PRI.
>
>     I am confused as to what approach to use to accomplish this.
>
>     Lua Script, Perl Script, Javascript, Phython, PHP, Ruby....etc.
>     Liverpie, ESL....etc.
>     Or a totally different approach.
>
>     I realize that there is not only one "correct" way to build this,
>     but I
>     am looking for advise from those who have done this type of thing
>     before.  I value the opinion of those who have gone down this road and
>     would be willing to share their thoughts on a recommended path to take
>     to accomplish this.
>
>     Thanks
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


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


From pjintheusa at gmail.com  Wed May 12 15:47:47 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Wed, 12 May 2010 18:47:47 -0400
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
In-Reply-To: <4BEB2B7D.4080406@gmail.com>
References: <0b27b9ae92e848c6edfcb0b6b20d255f@mail.gmail.com>
	<AANLkTiloS2j7QKkitjSmhECZXANvXnX8NH3fZiBL1qnm@mail.gmail.com>
	<4BEB2B7D.4080406@gmail.com>
Message-ID: <AANLkTil9T_6yFvKA8FuoicbvpeZOIkI0XRguUkgO25hL@mail.gmail.com>

Since you are on a Windows platform anyway - you could also look at
mod_managed, that would allow you build the application code in C#. The
access to your existing DB would potentially remain the same. It would be
quite easy to spin a monitor thread in C# and monitor a DB for new calls and
dispatch those calls etc.

We currently use Dialogic HMP in a Windows environment and our app has
ported well with out any change to the backend non telephony code.

On Wed, May 12, 2010 at 6:28 PM, Phone <phone.bytes at gmail.com> wrote:

> This is legit.  We are not into harassing anyone.  There are too many
> people doing that already.
>
> We currently have this working on a Windows/Dialogic platform, but would
> like to get it over to FreeSWITCH for a number of reasons.  It is always
> nice to avoid spending alot of time developing using a method that you
> later find has serious issues or limitations and then having to change
> horses and start over.  I just thought I would tap on someone else's
> wisdom in regards to a good way to talk to FS that works well.  I did
> notice that the wiki talks like LUA is the preferred way to go.
>
> Thanks
>
>
>
> Michael Collins wrote:
> >
> >
> > On Wed, May 12, 2010 at 2:50 PM, Kristian Kielhofner
> > <kris at kriskinc.com <mailto:kris at kriskinc.com>> wrote:
> >
> >     Are you going to be calling my cell phone to ask if I owe $10,000
> >     or more to
> >     the IRS?
> >
> >     Seriously, the world does not need another robodialer to harass
> >     people...
> >
> > Agreed, however there are legitimate use cases for this, like a
> > retailer calling to say an order is in, or reminder calls sent out by
> > a doctor's office. But yes, if it's dialing for dollars then I'm with
> > you: Por favor! No mas!
> > -MC
> >
> >
> >
> >     With that being said, write your app in whatever you want using
> >     ESL and
> >     connect to the socket to originate calls and listen for events.
> >
> >
> >     --
> >     Kristian Kielhofner
> >     http://blog.krisk.org
> >
> >     ----- Original Message -----
> >     From: freeswitch-users-bounces at lists.freeswitch.org
> >     <mailto:freeswitch-users-bounces at lists.freeswitch.org>
> >     <freeswitch-users-bounces at lists.freeswitch.org
> >     <mailto:freeswitch-users-bounces at lists.freeswitch.org>>
> >     To: freeswitch-users at lists.freeswitch.org
> >     <mailto:freeswitch-users at lists.freeswitch.org>
> >     <freeswitch-users at lists.freeswitch.org
> >     <mailto:freeswitch-users at lists.freeswitch.org>>
> >     Sent: Wed May 12 15:40:58 2010
> >     Subject: [Freeswitch-users] Questions on Building an application
> >     forFreeSWITCHI am looking for a boot in the right direction on the
> >     following project.
> >
> >     I am looking to build an "application" with FS where it will
> >     frequently
> >     check a database and then make an outbound call when a record with
> >     certain conditions is found in the DB.  It needs to determine if the
> >     call was then answered by a machine/vm, busy, or answered by a
> person,
> >     play a message and get a response back to be recorded in the db.
> >      It may
> >     find many calls that need to be made at the same time.  It will be
> >     running on a PRI.
> >
> >     I am confused as to what approach to use to accomplish this.
> >
> >     Lua Script, Perl Script, Javascript, Phython, PHP, Ruby....etc.
> >     Liverpie, ESL....etc.
> >     Or a totally different approach.
> >
> >     I realize that there is not only one "correct" way to build this,
> >     but I
> >     am looking for advise from those who have done this type of thing
> >     before.  I value the opinion of those who have gone down this road
> and
> >     would be willing to share their thoughts on a recommended path to
> take
> >     to accomplish this.
> >
> >     Thanks
> >
> >
> >     _______________________________________________
> >     FreeSWITCH-users mailing list
> >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     http://www.freeswitch.org
> >
> >     _______________________________________________
> >     FreeSWITCH-users mailing list
> >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     http://www.freeswitch.org
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Wed May 12 16:14:47 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 12 May 2010 18:14:47 -0500
Subject: [Freeswitch-users] ClueCon MMX in 3 months Register Early / Become
	a Sponsor
Message-ID: <AANLkTinyeCJxMIWGVhGephhiEjtC2rPPM_cLS2artfiH@mail.gmail.com>

This is the time of year we like to push for those interested in ClueCon to
register.
Knowing the registration count makes it easier to plan and guarantees that
everyone gets a room.

Also, we have a handful of speaker slots still available if anyone wants to
speak, contact marketing at cluecon.com

Finally, if your company is benefiting from open source software such as
FreeSWITCH or you have some new telephony products and services
and you want to have your logo engraved into our prizes and proudly
displayed on a banner at the conference,
contact marketing at cluecon.com as well to become a ClueCon Sponsor.

-- 
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
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From msc at freeswitch.org  Wed May 12 16:28:14 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 12 May 2010 16:28:14 -0700
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
In-Reply-To: <AANLkTil9T_6yFvKA8FuoicbvpeZOIkI0XRguUkgO25hL@mail.gmail.com>
References: <0b27b9ae92e848c6edfcb0b6b20d255f@mail.gmail.com>
	<AANLkTiloS2j7QKkitjSmhECZXANvXnX8NH3fZiBL1qnm@mail.gmail.com>
	<4BEB2B7D.4080406@gmail.com>
	<AANLkTil9T_6yFvKA8FuoicbvpeZOIkI0XRguUkgO25hL@mail.gmail.com>
Message-ID: <AANLkTikLpby-J0X6BvvutW7XxpgWylqZM3jU6SrREwBp@mail.gmail.com>

On Wed, May 12, 2010 at 3:47 PM, Phillip Jones <pjintheusa at gmail.com> wrote:

> Since you are on a Windows platform anyway - you could also look at
> mod_managed, that would allow you build the application code in C#. The
> access to your existing DB would potentially remain the same. It would be
> quite easy to spin a monitor thread in C# and monitor a DB for new calls and
> dispatch those calls etc.
>
> We currently use Dialogic HMP in a Windows environment and our app has
> ported well with out any change to the backend non telephony code.
>
> This is indeed one of the advantages of ESL and the event socket. If you
can abstract away the heavy lifting of the phone calls and handling from the
db/logic/etc. then it is much easier to port and to maintain.

In any case, use whatever language supports the event socket and is familiar
to you. In all honestly, I just use Perl. It is freakishly good at this sort
of thing, at least for me. ( http://xkcd.com/224/ )

-MC
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From gabe at gundy.org  Wed May 12 16:45:22 2010
From: gabe at gundy.org (Gabriel Gunderson)
Date: Wed, 12 May 2010 17:45:22 -0600
Subject: [Freeswitch-users] Memory leak in mod_nibblebill or in ODBC
	core?
In-Reply-To: <201005120659.04426.sos@sokhapkin.dyndns.org>
References: <201004170926.24979.sos@sokhapkin.dyndns.org>
	<201005120623.50868.sos@sokhapkin.dyndns.org>
	<AANLkTilzor0UZG3X6XsltRs_uBGkR3aqmOSmhHZRIQvS@mail.gmail.com>
	<201005120659.04426.sos@sokhapkin.dyndns.org>
Message-ID: <AANLkTilonpmyZKHxwemVQrd_jkUsfoIt6khu7287iXf7@mail.gmail.com>

On Wed, May 12, 2010 at 4:59 AM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org> wrote:
> Correct, no more unbouded memory growth.

So, can we close this thread as "FIXED" ?     :^)

Gabe


From phone.bytes at gmail.com  Wed May 12 16:46:28 2010
From: phone.bytes at gmail.com (Phone)
Date: Wed, 12 May 2010 17:46:28 -0600
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
In-Reply-To: <490aedfdd7c8ac680bbe0d7f68c786de@mail.gmail.com>
References: <490aedfdd7c8ac680bbe0d7f68c786de@mail.gmail.com>
Message-ID: <4BEB3DD4.9070807@gmail.com>

Thanks for sharing your experience.

We would like to get over to linux with this.

I am still trying to get my head around how this generally works.  I 
guess we would write a client to sit and run on the linux box where it 
could periodically check the DB and then talk sockets to FS to make the 
calls?

Can you point me to examples of talking sockets?





Kristian Kielhofner wrote:
> Glad to hear.
>
> I still say to build the app and logic completely separately and connect it
> to FreeSWITCH over the socket using ESL.
>
>
> --
> Kristian Kielhofner
> http://blog.krisk.org
>
> ----- Original Message -----
> From: freeswitch-users-bounces at lists.freeswitch.org
> <freeswitch-users-bounces at lists.freeswitch.org>
> To: freeswitch-users at lists.freeswitch.org
> <freeswitch-users at lists.freeswitch.org>
> Sent: Wed May 12 18:28:13 2010
> Subject: Re: [Freeswitch-users] Questions on Building an application
> forFreeSWITCHThis is legit.  We are not into harassing anyone.  There
> are too many
> people doing that already.
>
> We currently have this working on a Windows/Dialogic platform, but would
> like to get it over to FreeSWITCH for a number of reasons.  It is always
> nice to avoid spending alot of time developing using a method that you
> later find has serious issues or limitations and then having to change
> horses and start over.  I just thought I would tap on someone else's
> wisdom in regards to a good way to talk to FS that works well.  I did
> notice that the wiki talks like LUA is the preferred way to go.
>
> Thanks
>
>
>
> Michael Collins wrote:
>   
>> On Wed, May 12, 2010 at 2:50 PM, Kristian Kielhofner
>> <kris at kriskinc.com <mailto:kris at kriskinc.com>> wrote:
>>
>>     Are you going to be calling my cell phone to ask if I owe $10,000
>>     or more to
>>     the IRS?
>>
>>     Seriously, the world does not need another robodialer to harass
>>     people...
>>
>> Agreed, however there are legitimate use cases for this, like a
>> retailer calling to say an order is in, or reminder calls sent out by
>> a doctor's office. But yes, if it's dialing for dollars then I'm with
>> you: Por favor! No mas!
>> -MC
>>
>>
>>
>>     With that being said, write your app in whatever you want using
>>     ESL and
>>     connect to the socket to originate calls and listen for events.
>>
>>
>>     --
>>     Kristian Kielhofner
>>     http://blog.krisk.org
>>
>>     ----- Original Message -----
>>     From: freeswitch-users-bounces at lists.freeswitch.org
>>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>
>>     <freeswitch-users-bounces at lists.freeswitch.org
>>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>>
>>     To: freeswitch-users at lists.freeswitch.org
>>     <mailto:freeswitch-users at lists.freeswitch.org>
>>     <freeswitch-users at lists.freeswitch.org
>>     <mailto:freeswitch-users at lists.freeswitch.org>>
>>     Sent: Wed May 12 15:40:58 2010
>>     Subject: [Freeswitch-users] Questions on Building an application
>>     forFreeSWITCHI am looking for a boot in the right direction on the
>>     following project.
>>
>>     I am looking to build an "application" with FS where it will
>>     frequently
>>     check a database and then make an outbound call when a record with
>>     certain conditions is found in the DB.  It needs to determine if the
>>     call was then answered by a machine/vm, busy, or answered by a person,
>>     play a message and get a response back to be recorded in the db.
>>      It may
>>     find many calls that need to be made at the same time.  It will be
>>     running on a PRI.
>>
>>     I am confused as to what approach to use to accomplish this.
>>
>>     Lua Script, Perl Script, Javascript, Phython, PHP, Ruby....etc.
>>     Liverpie, ESL....etc.
>>     Or a totally different approach.
>>
>>     I realize that there is not only one "correct" way to build this,
>>     but I
>>     am looking for advise from those who have done this type of thing
>>     before.  I value the opinion of those who have gone down this road and
>>     would be willing to share their thoughts on a recommended path to take
>>     to accomplish this.
>>
>>     Thanks
>>
>>
>>     _______________________________________________
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>>
>>     _______________________________________________
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>>
>>
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From phone.bytes at gmail.com  Wed May 12 16:49:23 2010
From: phone.bytes at gmail.com (Phone)
Date: Wed, 12 May 2010 17:49:23 -0600
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
In-Reply-To: <AANLkTikLpby-J0X6BvvutW7XxpgWylqZM3jU6SrREwBp@mail.gmail.com>
References: <0b27b9ae92e848c6edfcb0b6b20d255f@mail.gmail.com>	<AANLkTiloS2j7QKkitjSmhECZXANvXnX8NH3fZiBL1qnm@mail.gmail.com>	<4BEB2B7D.4080406@gmail.com>	<AANLkTil9T_6yFvKA8FuoicbvpeZOIkI0XRguUkgO25hL@mail.gmail.com>
	<AANLkTikLpby-J0X6BvvutW7XxpgWylqZM3jU6SrREwBp@mail.gmail.com>
Message-ID: <4BEB3E83.3040502@gmail.com>

Thanks MC,

Can you point me to any example Perl code to start from?




Michael Collins wrote:
>
>
> On Wed, May 12, 2010 at 3:47 PM, Phillip Jones <pjintheusa at gmail.com 
> <mailto:pjintheusa at gmail.com>> wrote:
>
>     Since you are on a Windows platform anyway - you could also look
>     at mod_managed, that would allow you build the application code in
>     C#. The access to your existing DB would potentially remain the
>     same. It would be quite easy to spin a monitor thread in C# and
>     monitor a DB for new calls and dispatch those calls etc.
>
>     We currently use Dialogic HMP in a Windows environment and our app
>     has ported well with out any change to the backend non telephony
>     code.
>
> This is indeed one of the advantages of ESL and the event socket. If 
> you can abstract away the heavy lifting of the phone calls and 
> handling from the db/logic/etc. then it is much easier to port and to 
> maintain.
>
> In any case, use whatever language supports the event socket and is 
> familiar to you. In all honestly, I just use Perl. It is freakishly 
> good at this sort of thing, at least for me. ( http://xkcd.com/224/ )
>
> -MC
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From edpimentl at gmail.com  Wed May 12 17:16:26 2010
From: edpimentl at gmail.com (EdPimentl)
Date: Wed, 12 May 2010 20:16:26 -0400
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
In-Reply-To: <4BEB3E83.3040502@gmail.com>
References: <0b27b9ae92e848c6edfcb0b6b20d255f@mail.gmail.com> 
	<AANLkTiloS2j7QKkitjSmhECZXANvXnX8NH3fZiBL1qnm@mail.gmail.com> 
	<4BEB2B7D.4080406@gmail.com>
	<AANLkTil9T_6yFvKA8FuoicbvpeZOIkI0XRguUkgO25hL@mail.gmail.com> 
	<AANLkTikLpby-J0X6BvvutW7XxpgWylqZM3jU6SrREwBp@mail.gmail.com> 
	<4BEB3E83.3040502@gmail.com>
Message-ID: <AANLkTimlNC1VlOk_HVqtECMtP8nwm9uVIMkbijlLHn_a@mail.gmail.com>

Why not REALLY go to the next level with an OffLine(stand alone) Browser
APP  or Google  Native CLIENT SDK (Javacript / HTML / CSS / and Native CODE)
http://elephantsquared.com/2010/05/12/nacl-google-released-a-developer-preview-of-the-native-client-sdk/
http://en.wikipedia.org/wiki/NPAPI

-E
http://vCardCloud.com




On Wed, May 12, 2010 at 7:49 PM, Phone <phone.bytes at gmail.com> wrote:

> Thanks MC,
>
> Can you point me to any example Perl code to start from?
>
>
>
>
> Michael Collins wrote:
> >
> >
> > On Wed, May 12, 2010 at 3:47 PM, Phillip Jones <pjintheusa at gmail.com
> > <mailto:pjintheusa at gmail.com>> wrote:
> >
> >     Since you are on a Windows platform anyway - you could also look
> >     at mod_managed, that would allow you build the application code in
> >     C#. The access to your existing DB would potentially remain the
> >     same. It would be quite easy to spin a monitor thread in C# and
> >     monitor a DB for new calls and dispatch those calls etc.
> >
> >     We currently use Dialogic HMP in a Windows environment and our app
> >     has ported well with out any change to the backend non telephony
> >     code.
> >
> > This is indeed one of the advantages of ESL and the event socket. If
> > you can abstract away the heavy lifting of the phone calls and
> > handling from the db/logic/etc. then it is much easier to port and to
> > maintain.
> >
> > In any case, use whatever language supports the event socket and is
> > familiar to you. In all honestly, I just use Perl. It is freakishly
> > good at this sort of thing, at least for me. ( http://xkcd.com/224/ )
> >
> > -MC
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tony.tin at noahmedia.com.hk  Wed May 12 19:11:57 2010
From: tony.tin at noahmedia.com.hk (Tony Tin)
Date: Thu, 13 May 2010 10:11:57 +0800
Subject: [Freeswitch-users] Call Forwarding
In-Reply-To: <AANLkTilGDHjAG7moKpERFGdD8lm-V2vjlbzLvpk0HQ4-@mail.gmail.com>
References: <AANLkTilGDHjAG7moKpERFGdD8lm-V2vjlbzLvpk0HQ4-@mail.gmail.com>
Message-ID: <AANLkTinnM9gLuCV27hARwMKHMjQ5svqfN5hGGJoW8Jvy@mail.gmail.com>

for PSTN number, below lines are working for me

<!-- the outbond caller id show on the called party's phone -->
<action application="set" data="effective_caller_id_number=29876543"/>
<!-- add this line to avoid missing dtmf tone -->
<action application="start_dtmf_generate" data="true"/>
<!-- make outbound call on second span 1st channel -->
<action application="bridge" data="OpenZAP/2/1/31234567"/>


Tony



On Thu, May 13, 2010 at 1:28 AM, Aloysius Lloyd <lloyd.aloysius at gmail.com>wrote:

> Hi All,
>
> How to Call Forward a Extension to another Extension or to a PSTN Number
>  in FreeSWITCH?
>
>
>
> Thanks
> Lloyd
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From djbinter at gmail.com  Wed May 12 21:39:11 2010
From: djbinter at gmail.com (DJB INTERNATIONAL)
Date: Wed, 12 May 2010 21:39:11 -0700
Subject: [Freeswitch-users] Question regarding UUID
Message-ID: <AANLkTimdfOlV4ZSY4OLmBGzZFzuC640IW596CHRhxPq6@mail.gmail.com>

I would like to know whether there is a possibility that FS could generate
the same UUID for a-leg if I restart FS often.

Thank you,
Dorn
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From rupa at rupa.com  Wed May 12 21:47:56 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 12 May 2010 23:47:56 -0500
Subject: [Freeswitch-users] Question regarding UUID
In-Reply-To: <AANLkTimdfOlV4ZSY4OLmBGzZFzuC640IW596CHRhxPq6@mail.gmail.com>
References: <AANLkTimdfOlV4ZSY4OLmBGzZFzuC640IW596CHRhxPq6@mail.gmail.com>
Message-ID: <AANLkTikq6cCnSvjaUWHzoX4sRS-NjOiLQK_fBrwFvEs8@mail.gmail.com>

no

On Wed, May 12, 2010 at 11:39 PM, DJB INTERNATIONAL <djbinter at gmail.com>wrote:

> I would like to know whether there is a possibility that FS could generate
> the same UUID for a-leg if I restart FS often.
>
> Thank you,
> Dorn
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From jan.berger at video24.no  Wed May 12 22:23:05 2010
From: jan.berger at video24.no (Jan Berger)
Date: Thu, 13 May 2010 07:23:05 +0200
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
In-Reply-To: <4BEB3DD4.9070807@gmail.com>
References: <490aedfdd7c8ac680bbe0d7f68c786de@mail.gmail.com>
	<4BEB3DD4.9070807@gmail.com>
Message-ID: <EC392AB972A94F8E8D8369C802771451@dell9400>

You don't need to pull a db these day's, you can use triggers to signal your
application - a bit depending on what db you use and what scripting they
offer beyond SQL. 

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Phone
Sent: 13. mai 2010 01:46
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Questions on Building an application for
FreeSWITCH

Thanks for sharing your experience.

We would like to get over to linux with this.

I am still trying to get my head around how this generally works.  I 
guess we would write a client to sit and run on the linux box where it 
could periodically check the DB and then talk sockets to FS to make the 
calls?

Can you point me to examples of talking sockets?





Kristian Kielhofner wrote:
> Glad to hear.
>
> I still say to build the app and logic completely separately and connect
it
> to FreeSWITCH over the socket using ESL.
>
>
> --
> Kristian Kielhofner
> http://blog.krisk.org
>
> ----- Original Message -----
> From: freeswitch-users-bounces at lists.freeswitch.org
> <freeswitch-users-bounces at lists.freeswitch.org>
> To: freeswitch-users at lists.freeswitch.org
> <freeswitch-users at lists.freeswitch.org>
> Sent: Wed May 12 18:28:13 2010
> Subject: Re: [Freeswitch-users] Questions on Building an application
> forFreeSWITCHThis is legit.  We are not into harassing anyone.  There
> are too many
> people doing that already.
>
> We currently have this working on a Windows/Dialogic platform, but would
> like to get it over to FreeSWITCH for a number of reasons.  It is always
> nice to avoid spending alot of time developing using a method that you
> later find has serious issues or limitations and then having to change
> horses and start over.  I just thought I would tap on someone else's
> wisdom in regards to a good way to talk to FS that works well.  I did
> notice that the wiki talks like LUA is the preferred way to go.
>
> Thanks
>
>
>
> Michael Collins wrote:
>   
>> On Wed, May 12, 2010 at 2:50 PM, Kristian Kielhofner
>> <kris at kriskinc.com <mailto:kris at kriskinc.com>> wrote:
>>
>>     Are you going to be calling my cell phone to ask if I owe $10,000
>>     or more to
>>     the IRS?
>>
>>     Seriously, the world does not need another robodialer to harass
>>     people...
>>
>> Agreed, however there are legitimate use cases for this, like a
>> retailer calling to say an order is in, or reminder calls sent out by
>> a doctor's office. But yes, if it's dialing for dollars then I'm with
>> you: Por favor! No mas!
>> -MC
>>
>>
>>
>>     With that being said, write your app in whatever you want using
>>     ESL and
>>     connect to the socket to originate calls and listen for events.
>>
>>
>>     --
>>     Kristian Kielhofner
>>     http://blog.krisk.org
>>
>>     ----- Original Message -----
>>     From: freeswitch-users-bounces at lists.freeswitch.org
>>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>
>>     <freeswitch-users-bounces at lists.freeswitch.org
>>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>>
>>     To: freeswitch-users at lists.freeswitch.org
>>     <mailto:freeswitch-users at lists.freeswitch.org>
>>     <freeswitch-users at lists.freeswitch.org
>>     <mailto:freeswitch-users at lists.freeswitch.org>>
>>     Sent: Wed May 12 15:40:58 2010
>>     Subject: [Freeswitch-users] Questions on Building an application
>>     forFreeSWITCHI am looking for a boot in the right direction on the
>>     following project.
>>
>>     I am looking to build an "application" with FS where it will
>>     frequently
>>     check a database and then make an outbound call when a record with
>>     certain conditions is found in the DB.  It needs to determine if the
>>     call was then answered by a machine/vm, busy, or answered by a
person,
>>     play a message and get a response back to be recorded in the db.
>>      It may
>>     find many calls that need to be made at the same time.  It will be
>>     running on a PRI.
>>
>>     I am confused as to what approach to use to accomplish this.
>>
>>     Lua Script, Perl Script, Javascript, Phython, PHP, Ruby....etc.
>>     Liverpie, ESL....etc.
>>     Or a totally different approach.
>>
>>     I realize that there is not only one "correct" way to build this,
>>     but I
>>     am looking for advise from those who have done this type of thing
>>     before.  I value the opinion of those who have gone down this road
and
>>     would be willing to share their thoughts on a recommended path to
take
>>     to accomplish this.
>>
>>     Thanks
>>
>>
>>     _______________________________________________
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>>
>>     _______________________________________________
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>>
>>
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


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



From babak.freeswitch at gmail.com  Wed May 12 22:58:08 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Thu, 13 May 2010 10:28:08 +0430
Subject: [Freeswitch-users] sqlite in modmanaged
Message-ID: <AANLkTinKrCXQpJoQC1DdOcKDAB-0NRmepzLApLO4vmvK@mail.gmail.com>

Hi
How can I send sql commands (querying about sip registrations) to sqlite in
mod_managed?
I know I can use "sofia status profile internal" using api interface but
when there are many registrations and I just wanna check some of them this
is not efficient
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From shaheryarkh at googlemail.com  Wed May 12 23:04:01 2010
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Thu, 13 May 2010 11:04:01 +0500
Subject: [Freeswitch-users] FreeSWITCH startup failure
In-Reply-To: <AANLkTimOPHv6StrrobNll6HbzV_FqdQrLE1i6zidqiW2@mail.gmail.com>
References: <AANLkTinuEQVMWopPUrCYJSQa2XHaMHVASI882_aH8k71@mail.gmail.com>
	<AANLkTimOPHv6StrrobNll6HbzV_FqdQrLE1i6zidqiW2@mail.gmail.com>
Message-ID: <AANLkTimsixBqQIBgE-oo8ya5v7VDPMcAS0GvKF_shMng@mail.gmail.com>

First i tried odbc but it gave this error. Then i removed everything (FS
binaries, configs, sources etc.), downloaded FS again and compiled it
without odbc support but again same problem. I can see sqlite files being
created by freeswitch in /usr/local/freeswitch/db folder but for some
reason, FS still can't find any table in it. The problem is not with just
interface table but all tables, include table related to sofia sip, voice
mail etc. etc.

I am using Ubuntu 10.04 32bit desktop edition, on Intel x86 with following
freeswitch configuration,

./configure --enable-optimization --disable-debug --enable-visibility
--enable-zrtp --enable-core-odbc-support --with-openssl
--with-java=/usr/lib/jvm/java-6-sun --with-python --with-erlang

FS compiles without any problem as all dependencies are already
pre-installed. But after installation when i start it with following options
as root user,

./freeswitch -hp -waste -core

The console soon become full with switch_core_sqldb.c errors and system gets
to slow down till it hangs.

I have tried different combinations of configure flags (e.g. removing zrtp
and odbc support etc.) but no success.

I have examined the sqlite files created by FS but it seems each file
contains only one table, while i suspect there should be more then one
tables, for example for sofia sip i remember there is a table for sip
registrations, sip presence, sip dialog etc. etc. but when i open
/usr/local/freeswitch/db/sofia_reg_internal.db file in SQLite manager, there
is only one empty table named sip_recovery.

Same thing happens when using odbc, only one table sip_recovery gets created
for sofia sip, the FS continues to complain that it can't find any table.

Please help as i have wasted 2 days without any luck on this.

Thank you.


On Wed, May 12, 2010 at 6:49 PM, Rupa Schomaker <rupa at rupa.com> wrote:

> THat commit just removes a .rej file, don't see how it can affect
>
> $ git diff
> --stat 1d2c64d33bf033788c5521c212b861ba11c229af..5530de66ef6880621086c6276731504459eb6709
>
> (those are the two SHA1s I'm getting stats on,  first is tony's just before
> my changes, second is mine)
>
>  src/include/switch_types.h.orig                    | 1714 ------
>  .../mod_conference/mod_conference.c.orig           | 6343
> --------------------
>  .../mod_conference/mod_conference.c.rej            |   50 -
>  src/mod/formats/mod_sndfile/mod_sndfile.c.orig     |  462 --
>  .../languages/mod_python/mod_python_wrap.cpp.rej   |   38 -
>  src/switch_core.c.orig                             | 2018 -------
>  src/switch_time.c.orig                             | 1962 ------
>  7 files changed, 0 insertions(+), 12587 deletions(-)
>
>
> Don't see how removing patch droppings can possibly cause your problem.
>
> The real issue is that the interfaces table doesn't exist.  The last error
> is because the table didn't exist for the records to be inserted into.  As
> with all core tables, this SHOULD be auto-created.
>
> Are you running against odbc for core or sqlite?
>
> On Wed, May 12, 2010 at 6:48 AM, Muhammad Shahzad <
> shaheryarkh at googlemail.com> wrote:
>
>> Hi,
>>
>> For last two days i am trying to setup a freeswitch box for an voip
>> application testing against it. FreeSWITCH compiles successfully but when i
>> try to start it i get a lot critical errors from switch_core_sqldb.c, e.g.
>>
>>
>> 2010-05-12 16:35:42.691358 [ERR] switch_core_sqldb.c:404 SQL ERR [no such
>> table: interfaces]
>> insert into interfaces
>> (type,name,description,syntax,ikey,filename,hostname)
>> values('chat','event','','','mod_dptools','/usr/local/freeswitch/mod/mod_dptools.so','lenovo');
>> insert into interfaces
>> (type,name,description,syntax,ikey,filename,hostname)
>> values('chat','api','','','mod_dptools','/usr/local/freeswitch/mod/mod_dptools.so','lenovo');
>> insert into interfaces
>> (type,name,description,syntax,ikey,filename,hostname)
>> values('api','expr','Eval an
>> expression','<expr>','mod_expr','/usr/local/freeswitch/mod/mod_expr.so','lenovo');
>>
>> 2010-05-12 16:35:42.691419 [ERR] switch_core_sqldb.c:670 SQL ERR [no such
>> table: interfaces]
>> 2010-05-12 16:35:42.791490 [CRIT] switch_core_sqldb.c:889 SQL thread
>> unable to commit transaction, records lost!
>>
>> and it continues till my machine eventually hangs. Its happening with
>> latest git revision, i don't know how to get revision number in git (i am
>> not much familiar with it) but "git log" command give this as first line.
>>
>> commit 5530de66ef6880621086c6276731504459eb6709
>> Author: Rupa Schomaker <rupa at rupa.com>
>> Date:   Wed May 12 03:32:18 2010 -0500
>>
>> I suspect the last line, i.e.
>> 2010-05-12 16:35:42.791490 [CRIT] switch_core_sqldb.c:889 SQL thread
>> unable to commit transaction, records lost!
>>
>> is the reason for these errors but i am not sure why its happening.
>>
>> Thank you.
>>
>>
>> --
>> Muhammad Shahzad
>> -----------------------------------
>> CISCO Rich Media Communication Specialist (CRMCS)
>> CISCO Certified Network Associate (CCNA)
>> Cell: +92 334 422 40 88
>> MSN: shari_786pk at hotmail.com
>> Email: shaheryarkh at googlemail.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From freeswitch.org at todandlorna.com  Wed May 12 23:30:39 2010
From: freeswitch.org at todandlorna.com (Tod Hansmann)
Date: Thu, 13 May 2010 00:30:39 -0600
Subject: [Freeswitch-users] sqlite in modmanaged
In-Reply-To: <AANLkTinKrCXQpJoQC1DdOcKDAB-0NRmepzLApLO4vmvK@mail.gmail.com>
References: <AANLkTinKrCXQpJoQC1DdOcKDAB-0NRmepzLApLO4vmvK@mail.gmail.com>
Message-ID: <4BEB9C8F.6000206@todandlorna.com>

Depending on your system, you can use either the Mono.Data.SqliteClient 
or get the System.Data.SQLite library from 
http://sqlite.phxsoftware.com/.  The former for linux, the latter for 
windows.

The sanity of this request is not something I'm familiar with, but that 
should enable you to take matters into your own hands, for good or ill =cP

-Tod Hansmann


On 5/12/2010 11:58 PM, babak yakhchali wrote:
> Hi
> How can I send sql commands (querying about sip registrations) to 
> sqlite in mod_managed?
> I know I can use "sofia status profile internal" using api interface 
> but when there are many registrations and I just wanna check some of 
> them this is not efficient
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    
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From babak.freeswitch at gmail.com  Thu May 13 00:51:53 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Thu, 13 May 2010 12:21:53 +0430
Subject: [Freeswitch-users] sqlite in modmanaged
In-Reply-To: <4BEB9C8F.6000206@todandlorna.com>
References: <AANLkTinKrCXQpJoQC1DdOcKDAB-0NRmepzLApLO4vmvK@mail.gmail.com>
	<4BEB9C8F.6000206@todandlorna.com>
Message-ID: <AANLkTim8FGWoVY2j3iZdU4n-8lIi8mcktJz6encL8L6R@mail.gmail.com>

u mean this is not a good idea to get registrations from sqlite??
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From freeswitch at gilligan.id.au  Thu May 13 02:32:50 2010
From: freeswitch at gilligan.id.au (Chris)
Date: Thu, 13 May 2010 19:32:50 +1000
Subject: [Freeswitch-users] mod_managed or a script,
	sending DTMF and receiving DTMF
Message-ID: <AANLkTikSTwOw1yiBGO9VrVDnsKtH7ySz8V2LxL1XUdru@mail.gmail.com>

HI,
I am currently working on a project in mod_managed and i am trying to
discover the best way to meet my requirements.  i am hoping someone will
have some ideas.  This could be implemented in one of the other scripting
language if needed.

What i am trying to do is reprogram a remote device via the phone.  The
device takes commands in the form of DTMF tones and responds in different
DTMF tones depending on success or failure. An example of the flow would be

freeswitch -> 342523#
device -> 1
freeswitch -> 356789#
device ->2

device always responds with one digit and freeswitch sends many.

I was hoping to do this with a database and mod_managed but i can't workout
how to send the DTMF in mod_managed unless i user audio files for it which
seems to be the wrong way to go to me. While i would prefer a mod_managed
solution i will take anything i can find.



Chris
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From tculjaga at gmail.com  Thu May 13 02:36:45 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 13 May 2010 11:36:45 +0200
Subject: [Freeswitch-users] fetch caller profile variables
Message-ID: <AANLkTinXlIUddb_xAWh1u3DLpAuuWQhisOOA8ygWLGnU@mail.gmail.com>

hello,

i need to fetch a caller profile variable (e.g. answer_stamp) from a
dialplan, can anyone help?


i tried with:

<action application="set" data="TEST=answer_stamp"/>
<action inline="true" application="set" data="TEST=answer_stamp"/>
<action inline="true" application="set" data="TEST=${answer_stamp}"/>


but it doesn't help...
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From tculjaga at gmail.com  Thu May 13 02:38:02 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Thu, 13 May 2010 11:38:02 +0200
Subject: [Freeswitch-users] fetch caller profile variables
In-Reply-To: <AANLkTinXlIUddb_xAWh1u3DLpAuuWQhisOOA8ygWLGnU@mail.gmail.com>
References: <AANLkTinXlIUddb_xAWh1u3DLpAuuWQhisOOA8ygWLGnU@mail.gmail.com>
Message-ID: <AANLkTikgPP1_nRzU6IgaM4AZIqO4q0Cx5trTVuLuPwiQ@mail.gmail.com>

also, how do i acced this variables from the dialplan ?


http://wiki.freeswitch.org/index.php?title=Category:Variable&until=Variable+ivr+menu+status


Regards,
Tihomir.




On Thu, May 13, 2010 at 11:36 AM, Tihomir Culjaga <tculjaga at gmail.com>wrote:

> hello,
>
> i need to fetch a caller profile variable (e.g. answer_stamp) from a
> dialplan, can anyone help?
>
>
> i tried with:
>
> <action application="set" data="TEST=answer_stamp"/>
> <action inline="true" application="set" data="TEST=answer_stamp"/>
> <action inline="true" application="set" data="TEST=${answer_stamp}"/>
>
>
> but it doesn't help...
>
>
>
>
>
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From lakindia89 at gmail.com  Thu May 13 05:45:17 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Thu, 13 May 2010 18:15:17 +0530
Subject: [Freeswitch-users] Alarm Cleared on channel - Suppressing
Message-ID: <AANLkTin4TqPxCPv3_aBywRGtFFncaDGMNfffC_g05i_D@mail.gmail.com>

Hi all,
I'm having sangoma A102 card and I'm using freetdm.
For every 5 minutes, I get "Alarm Cleared on channel 1:1 to 1:31" and in
FreeSwitch console.

One of my friend told that in Asterisk, he will suppress  this by setting
resetinterval=never .
Can any one please tell when this will occur and is there any way in
FreeSwitch to suppress it?? please help!.
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From lloyd.aloysius at gmail.com  Thu May 13 05:46:23 2010
From: lloyd.aloysius at gmail.com (Aloysius Lloyd)
Date: Thu, 13 May 2010 08:46:23 -0400
Subject: [Freeswitch-users] Gateway Registration Issues
Message-ID: <AANLkTim-d2_nD3lLS3YlqciwrgJpGP5jMyCfSWtxI_Ly@mail.gmail.com>

Hi All,

I am trying to connect IRISTEL ( CLEC ) in Canada and FreeSWITCH. I have the
following gateway configuration

<include>
  <gateway name="wwc_test">
    <param name="auth-username" value="username"/>
    <param name="username" value="14161231234" />
    <param name="password" value="password" />
    <param name="proxy" value="sbc1.iristel.net" />
    <param name="realm" value="iristel.net" />
    <param name="expire-seconds" value="120"/>
    <param name="register" value="true"/>
    <param name="retry-seconds" value="30"/>
    <param name="ping" value="25"/>
  </gateway>
</include>


The above configuration is not working. I have the following error in cli
Error.

*2010-05-13 08:34:15.987465 [ERR] sofia_reg.c:1501 14161231234 Registration
Failed with status Conflict [409]. failure #1*

====

Also carrier saying there is conflict in the contact field.Here is how
they receive now.

*Contact: <sip:gw+test_iristel at 10.20.30.1:5080
;transport=udp;gw=test_iristel>.*

But the contact field should be like below

Contact: <14161231234 at 10.20.30.1:5080;

Any suggestions?

Thanks in advance.

Lloyd
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From pjintheusa at gmail.com  Thu May 13 05:50:49 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Thu, 13 May 2010 08:50:49 -0400
Subject: [Freeswitch-users] mod_managed or a script,
	sending DTMF and 	receiving DTMF
In-Reply-To: <AANLkTikSTwOw1yiBGO9VrVDnsKtH7ySz8V2LxL1XUdru@mail.gmail.com>
References: <AANLkTikSTwOw1yiBGO9VrVDnsKtH7ySz8V2LxL1XUdru@mail.gmail.com>
Message-ID: <AANLkTik1BYHsuHlfS9FAD3pEGS5GaEIINiiY9708sFji@mail.gmail.com>

Try using http://wiki.freeswitch.org/wiki/Mod_commands#uuid_send_dtmf

On Thu, May 13, 2010 at 5:32 AM, Chris <freeswitch at gilligan.id.au> wrote:

> HI,
> I am currently working on a project in mod_managed and i am trying to
> discover the best way to meet my requirements.  i am hoping someone will
> have some ideas.  This could be implemented in one of the other scripting
> language if needed.
>
> What i am trying to do is reprogram a remote device via the phone.  The
> device takes commands in the form of DTMF tones and responds in different
> DTMF tones depending on success or failure. An example of the flow would be
>
> freeswitch -> 342523#
> device -> 1
> freeswitch -> 356789#
> device ->2
>
> device always responds with one digit and freeswitch sends many.
>
> I was hoping to do this with a database and mod_managed but i can't workout
> how to send the DTMF in mod_managed unless i user audio files for it which
> seems to be the wrong way to go to me. While i would prefer a mod_managed
> solution i will take anything i can find.
>
>
>
> Chris
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From pjintheusa at gmail.com  Thu May 13 05:57:25 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Thu, 13 May 2010 08:57:25 -0400
Subject: [Freeswitch-users] sqlite in modmanaged
In-Reply-To: <AANLkTim8FGWoVY2j3iZdU4n-8lIi8mcktJz6encL8L6R@mail.gmail.com>
References: <AANLkTinKrCXQpJoQC1DdOcKDAB-0NRmepzLApLO4vmvK@mail.gmail.com>
	<4BEB9C8F.6000206@todandlorna.com>
	<AANLkTim8FGWoVY2j3iZdU4n-8lIi8mcktJz6encL8L6R@mail.gmail.com>
Message-ID: <AANLkTik9QO4BIhurIIF6nZp9eF0uVmeM7dlmlV6MTN6y@mail.gmail.com>

I use the same method to access registrations, (albeit in a MySQL DB through
ODBC)

It works fine. The problem I worry about - is scalability - as I have it
today in my prototype - I am building and tearing down an ODBC connection on
every query. This is not desirable, so I have to pool the connection in some
way. When that is done, I can not see a reason not to do it this way?

On Thu, May 13, 2010 at 3:51 AM, babak yakhchali <babak.freeswitch at gmail.com
> wrote:

> u mean this is not a good idea to get registrations from sqlite??
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at gmail.com  Thu May 13 05:58:04 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Thu, 13 May 2010 14:58:04 +0200
Subject: [Freeswitch-users] Gateway Registration Issues
In-Reply-To: <AANLkTim-d2_nD3lLS3YlqciwrgJpGP5jMyCfSWtxI_Ly@mail.gmail.com>
References: <AANLkTim-d2_nD3lLS3YlqciwrgJpGP5jMyCfSWtxI_Ly@mail.gmail.com>
Message-ID: <983B7F89-A86E-4AFE-91E6-553D1F0B6495@gmail.com>

Aloysius,,

That is an old-skool carrier.
You have to add this line to your gateway params:

<param name="extension-in-contact" value="true"/>

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 13/05/2010 ? 14:46, Aloysius Lloyd a ?crit :

> Hi All,
>
> I am trying to connect IRISTEL ( CLEC ) in Canada and FreeSWITCH. I  
> have the following gateway configuration
>
> <include>
>   <gateway name="wwc_test">
>     <param name="auth-username" value="username"/>
>     <param name="username" value="14161231234" />
>     <param name="password" value="password" />
>     <param name="proxy" value="sbc1.iristel.net" />
>     <param name="realm" value="iristel.net" />
>     <param name="expire-seconds" value="120"/>
>     <param name="register" value="true"/>
>     <param name="retry-seconds" value="30"/>
>     <param name="ping" value="25"/>
>   </gateway>
> </include>
>
>
> The above configuration is not working. I have the following error  
> in cli Error.
>
> 2010-05-13 08:34:15.987465 [ERR] sofia_reg.c:1501 14161231234  
> Registration Failed with status Conflict [409]. failure #1
>
> ====
>
> Also carrier saying there is conflict in the contact field.Here is  
> how they receive now.
>
> Contact: <sip:gw+test_iristel at 10.20.30.1:5080;transport=udp;gw=test_iristel 
> >.
>
> But the contact field should be like below
>
> Contact: <14161231234 at 10.20.30.1:5080;
>
> Any suggestions?
>
> Thanks in advance.
>
> Lloyd
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From phunk0000 at hotmail.com  Thu May 13 06:11:18 2010
From: phunk0000 at hotmail.com (Todd)
Date: Thu, 13 May 2010 09:11:18 -0400
Subject: [Freeswitch-users] mod_limit
Message-ID: <BLU110-DS61D013776D8066F88BE5EA7FC0@phx.gbl>

Hey list, could someone please help me out with the syntax for mod_limit?  I
want to use it to block any CID that call us more than 20 time in 5 mintues.
I found this in the wiki:

 

<action application="limit_hash" data="${sip_received_ip}
${destination_number} 5/600" />

 

And I am sure it's close, but I need some help with the variable names and
exact syntax to block inbound caller ID numbers that violate the threshold
parameters of 20 times in 5 minutes.  Thanks List!

 

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From info at evestech.com  Thu May 13 07:04:08 2010
From: info at evestech.com (Kashif Kahn)
Date: Thu, 13 May 2010 07:04:08 -0700 (PDT)
Subject: [Freeswitch-users] Robust Affordable Speech Recognition
Message-ID: <600175.94339.qm@web208.biz.mail.re2.yahoo.com>

Dear All,

All those who have wanted a speech recognition solution for Freeswitch but found the software cost too expensive or the 
recognition accuracy unsatisfactory, please consider Vestec 
Speech Engine for Freeswitchat: http://www.vestec.ca/products A starter kit - which is a specially priced one port (ie. one 
channel) license for the standard engine - is available for only $25. 
Additional ports (channels) licenses can be purchased for $99/port. The engine comes with a free-of-charge Freeswitch connector, thereby allowing direct interaction via 
Dialplan.

Best regards,
-Kashif
 Kashif Kahn
VP, Business Development
Vestec, Inc.
Waterloo, ON Canada
phone: (519) 885-7615
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From garrison at codefix.net  Thu May 13 07:37:53 2010
From: garrison at codefix.net (Garrison Hoffman)
Date: Thu, 13 May 2010 10:37:53 -0400
Subject: [Freeswitch-users] Recommendations for adapter replacement ...
Message-ID: <1273761473.1407.26.camel@strontium>

I'm finally convinced my Grandstream HT386 is too much trouble. There's
got to be some clever engineering in a device which manages to have a
problem with every FS configuration I've tested.

What I need now is another dual FXS adapter, one port is for a cordless
phone and the other is a dedicated fax line.

I'm considering an unlocked Linksys PAP2T, and a Grandstream HT502 but
I'm open to other recommendations. The HT502 is only on my list due to
UPnP & TLS support and because my GX2020 has always worked well.

Both products are available at voipsupply.com for under $60, although I
have no fixed price cutoff.

-gh



From anthony.minessale at gmail.com  Thu May 13 07:39:26 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 13 May 2010 09:39:26 -0500
Subject: [Freeswitch-users] fetch caller profile variables
In-Reply-To: <AANLkTinXlIUddb_xAWh1u3DLpAuuWQhisOOA8ygWLGnU@mail.gmail.com>
References: <AANLkTinXlIUddb_xAWh1u3DLpAuuWQhisOOA8ygWLGnU@mail.gmail.com>
Message-ID: <AANLkTim4KO8XRwl-n9-tq-BSFpwC2Jds1H752Rh9uNIk@mail.gmail.com>

answer_stamp is a regular channel variable not a caller profile variable
${answer_stamp} is correct but its only set once the channel is hungup

billing is best done in a separate process from the dialplan on the backend
via EVENTS or CDR


On Thu, May 13, 2010 at 4:36 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> hello,
>
> i need to fetch a caller profile variable (e.g. answer_stamp) from a
> dialplan, can anyone help?
>
>
> i tried with:
>
> <action application="set" data="TEST=answer_stamp"/>
> <action inline="true" application="set" data="TEST=answer_stamp"/>
> <action inline="true" application="set" data="TEST=${answer_stamp}"/>
>
>
> but it doesn't help...
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <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
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From anthony.minessale at gmail.com  Thu May 13 07:46:19 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 13 May 2010 09:46:19 -0500
Subject: [Freeswitch-users] FreeSWITCH startup failure
In-Reply-To: <AANLkTimsixBqQIBgE-oo8ya5v7VDPMcAS0GvKF_shMng@mail.gmail.com>
References: <AANLkTinuEQVMWopPUrCYJSQa2XHaMHVASI882_aH8k71@mail.gmail.com>
	<AANLkTimOPHv6StrrobNll6HbzV_FqdQrLE1i6zidqiW2@mail.gmail.com>
	<AANLkTimsixBqQIBgE-oo8ya5v7VDPMcAS0GvKF_shMng@mail.gmail.com>
Message-ID: <AANLkTilchtcc5k38-8TNm-a5SK0u30O4J76cxDFnAD57@mail.gmail.com>

who suggested those configure flags?
"-enable-optimization --disable-debug --enable-visibility" probably should
not be passed

regardless if you configure with odbc or not it will not use it unles you
configure it to.
It will always us sqlite.

You are using bleeding edge distro which is an at-your-own risk decision.

you probably need to rm -f /usr/local/freeswitch before you re-install from
scratch.

Basically you are wasting your 2 days by not following
the recommendations we lay out for people who are unsure how to properly
configure their system.  Learn to take the tao approach and "go with the
flow" and you could spend those 2 days in a hammock.




On Thu, May 13, 2010 at 1:04 AM, Muhammad Shahzad <
shaheryarkh at googlemail.com> wrote:

> First i tried odbc but it gave this error. Then i removed everything (FS
> binaries, configs, sources etc.), downloaded FS again and compiled it
> without odbc support but again same problem. I can see sqlite files being
> created by freeswitch in /usr/local/freeswitch/db folder but for some
> reason, FS still can't find any table in it. The problem is not with just
> interface table but all tables, include table related to sofia sip, voice
> mail etc. etc.
>
> I am using Ubuntu 10.04 32bit desktop edition, on Intel x86 with following
> freeswitch configuration,
>
> ./configure --enable-optimization --disable-debug --enable-visibility
> --enable-zrtp --enable-core-odbc-support --with-openssl
> --with-java=/usr/lib/jvm/java-6-sun --with-python --with-erlang
>
> FS compiles without any problem as all dependencies are already
> pre-installed. But after installation when i start it with following options
> as root user,
>
> ./freeswitch -hp -waste -core
>
> The console soon become full with switch_core_sqldb.c errors and system
> gets to slow down till it hangs.
>
> I have tried different combinations of configure flags (e.g. removing zrtp
> and odbc support etc.) but no success.
>
> I have examined the sqlite files created by FS but it seems each file
> contains only one table, while i suspect there should be more then one
> tables, for example for sofia sip i remember there is a table for sip
> registrations, sip presence, sip dialog etc. etc. but when i open
> /usr/local/freeswitch/db/sofia_reg_internal.db file in SQLite manager, there
> is only one empty table named sip_recovery.
>
> Same thing happens when using odbc, only one table sip_recovery gets
> created for sofia sip, the FS continues to complain that it can't find any
> table.
>
> Please help as i have wasted 2 days without any luck on this.
>
> Thank you.
>
>
>
> On Wed, May 12, 2010 at 6:49 PM, Rupa Schomaker <rupa at rupa.com> wrote:
>
>> THat commit just removes a .rej file, don't see how it can affect
>>
>> $ git diff
>> --stat 1d2c64d33bf033788c5521c212b861ba11c229af..5530de66ef6880621086c6276731504459eb6709
>>
>> (those are the two SHA1s I'm getting stats on,  first is tony's just
>> before my changes, second is mine)
>>
>>  src/include/switch_types.h.orig                    | 1714 ------
>>  .../mod_conference/mod_conference.c.orig           | 6343
>> --------------------
>>  .../mod_conference/mod_conference.c.rej            |   50 -
>>  src/mod/formats/mod_sndfile/mod_sndfile.c.orig     |  462 --
>>  .../languages/mod_python/mod_python_wrap.cpp.rej   |   38 -
>>  src/switch_core.c.orig                             | 2018 -------
>>   src/switch_time.c.orig                             | 1962 ------
>>  7 files changed, 0 insertions(+), 12587 deletions(-)
>>
>>
>> Don't see how removing patch droppings can possibly cause your problem.
>>
>> The real issue is that the interfaces table doesn't exist.  The last error
>> is because the table didn't exist for the records to be inserted into.  As
>> with all core tables, this SHOULD be auto-created.
>>
>> Are you running against odbc for core or sqlite?
>>
>> On Wed, May 12, 2010 at 6:48 AM, Muhammad Shahzad <
>> shaheryarkh at googlemail.com> wrote:
>>
>>> Hi,
>>>
>>> For last two days i am trying to setup a freeswitch box for an voip
>>> application testing against it. FreeSWITCH compiles successfully but when i
>>> try to start it i get a lot critical errors from switch_core_sqldb.c, e.g.
>>>
>>>
>>> 2010-05-12 16:35:42.691358 [ERR] switch_core_sqldb.c:404 SQL ERR [no such
>>> table: interfaces]
>>> insert into interfaces
>>> (type,name,description,syntax,ikey,filename,hostname)
>>> values('chat','event','','','mod_dptools','/usr/local/freeswitch/mod/mod_dptools.so','lenovo');
>>> insert into interfaces
>>> (type,name,description,syntax,ikey,filename,hostname)
>>> values('chat','api','','','mod_dptools','/usr/local/freeswitch/mod/mod_dptools.so','lenovo');
>>> insert into interfaces
>>> (type,name,description,syntax,ikey,filename,hostname)
>>> values('api','expr','Eval an
>>> expression','<expr>','mod_expr','/usr/local/freeswitch/mod/mod_expr.so','lenovo');
>>>
>>> 2010-05-12 16:35:42.691419 [ERR] switch_core_sqldb.c:670 SQL ERR [no such
>>> table: interfaces]
>>> 2010-05-12 16:35:42.791490 [CRIT] switch_core_sqldb.c:889 SQL thread
>>> unable to commit transaction, records lost!
>>>
>>> and it continues till my machine eventually hangs. Its happening with
>>> latest git revision, i don't know how to get revision number in git (i am
>>> not much familiar with it) but "git log" command give this as first line.
>>>
>>> commit 5530de66ef6880621086c6276731504459eb6709
>>> Author: Rupa Schomaker <rupa at rupa.com>
>>> Date:   Wed May 12 03:32:18 2010 -0500
>>>
>>> I suspect the last line, i.e.
>>> 2010-05-12 16:35:42.791490 [CRIT] switch_core_sqldb.c:889 SQL thread
>>> unable to commit transaction, records lost!
>>>
>>> is the reason for these errors but i am not sure why its happening.
>>>
>>> Thank you.
>>>
>>>
>>> --
>>> Muhammad Shahzad
>>> -----------------------------------
>>> CISCO Rich Media Communication Specialist (CRMCS)
>>> CISCO Certified Network Associate (CCNA)
>>> Cell: +92 334 422 40 88
>>> MSN: shari_786pk at hotmail.com
>>> Email: shaheryarkh at googlemail.com
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Muhammad Shahzad
> -----------------------------------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +92 334 422 40 88
> MSN: shari_786pk at hotmail.com
> Email: shaheryarkh at googlemail.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <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
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From freeswitch.org at todandlorna.com  Thu May 13 07:53:16 2010
From: freeswitch.org at todandlorna.com (Tod Hansmann)
Date: Thu, 13 May 2010 08:53:16 -0600
Subject: [Freeswitch-users] sqlite in modmanaged
In-Reply-To: <AANLkTim8FGWoVY2j3iZdU4n-8lIi8mcktJz6encL8L6R@mail.gmail.com>
References: <AANLkTinKrCXQpJoQC1DdOcKDAB-0NRmepzLApLO4vmvK@mail.gmail.com>	<4BEB9C8F.6000206@todandlorna.com>
	<AANLkTim8FGWoVY2j3iZdU4n-8lIi8mcktJz6encL8L6R@mail.gmail.com>
Message-ID: <4BEC125C.1020009@todandlorna.com>

I have absolutely no idea.  It could be fine, it could be problematic.  
I just didn't want to endorse it, just in case =c)

-Tod Hansmann

On 5/13/2010 1:51 AM, babak yakhchali wrote:
> u mean this is not a good idea to get registrations from sqlite?? 


From david.ponzone at gmail.com  Thu May 13 08:36:47 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Thu, 13 May 2010 17:36:47 +0200
Subject: [Freeswitch-users] mod_limit
In-Reply-To: <BLU110-DS61D013776D8066F88BE5EA7FC0@phx.gbl>
References: <BLU110-DS61D013776D8066F88BE5EA7FC0@phx.gbl>
Message-ID: <541072BF-8F7E-4711-AF1C-2BD391D02CEA@gmail.com>

Todd,

it would be something like:

<action application="limit_hash" data="whatever ${caller_id_number}  
20/300" />

For "whatever", you can use a fixed string, if you want your limit to  
be global, or else you can use another variable you think is valid (if  
you want the limit to be specific for one peer for instance, you can  
use network_addr).

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 13/05/2010 ? 15:11, Todd a ?crit :

> Hey list, could someone please help me out with the syntax for  
> mod_limit?  I want to use it to block any CID that call us more than  
> 20 time in 5 mintues.  I found this in the wiki:
>
> <action application="limit_hash" data="${sip_received_ip} $ 
> {destination_number} 5/600" />
>
> And I am sure it?s close, but I need some help with the variable  
> names and exact syntax to block inbound caller ID numbers that  
> violate the threshold parameters of 20 times in 5 minutes.  Thanks  
> List!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jan.berger at video24.no  Thu May 13 08:36:54 2010
From: jan.berger at video24.no (Jan Berger)
Date: Thu, 13 May 2010 17:36:54 +0200
Subject: [Freeswitch-users] Robust Affordable Speech Recognition
In-Reply-To: <600175.94339.qm@web208.biz.mail.re2.yahoo.com>
References: <600175.94339.qm@web208.biz.mail.re2.yahoo.com>
Message-ID: <FCC1A29F6BF74D48AB906CFC2BC18CA2@dell9400>

Hi,

 

Hope you don't mind a few nosy questions.

 

What languages do you support (1) national characters, (2) pre-build
databases.

 

What are your recognition stats per language?

 

What are the benefits of using this compared to Sphinx/Pocketsphinx?

 

How does the (1) installation of a license happen, (2) license check happen.

 

Does the licensing server support redundancy schemes? 

 

Jan

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kashif
Kahn
Sent: 13. mai 2010 16:04
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Robust Affordable Speech Recognition

 

Dear All,

All those who have wanted a speech recognition solution for Freeswitch but
found the software cost too expensive or the recognition accuracy
unsatisfactory, please consider Vestec Speech Engine for Freeswitch at:
http://www.vestec.ca/products A starter kit - which is a specially priced
one port (ie. one channel) license for the standard engine - is available
for only $25. Additional ports (channels) licenses can be purchased for
$99/port. The engine comes with a free-of-charge Freeswitch connector,
thereby allowing direct interaction via Dialplan.

Best regards,
-Kashif

 

Kashif Kahn
VP, Business Development
Vestec, Inc.
Waterloo, ON Canada
phone: (519) 885-7615

 

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From david.ponzone at gmail.com  Thu May 13 08:43:35 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Thu, 13 May 2010 17:43:35 +0200
Subject: [Freeswitch-users] Recommendations for adapter replacement ...
In-Reply-To: <1273761473.1407.26.camel@strontium>
References: <1273761473.1407.26.camel@strontium>
Message-ID: <14E11787-9AE2-4D06-A89C-EED6D0F3E55F@gmail.com>

If you want a fax on one line, you plan to use fax over G711 or T38 ?
As far as I know, the PAP2T does not have T38.
Fax over G711 is known to be quite unreliable.

You should look at Audiocodes and Patton.
Both of those vendors are known to produce good devices, although I  
remember someone saying that the analog devices from Audiocodes were  
not as good as the digital ones.

If you really need fax, even T38, you should not have a fixed price  
cutoff.
You should rather try all the boxes on the market, and if you'are  
lucky enough to find one working with your gateway (not with FS, FS is  
transparent for that), stay with that and hope the vendor won't stop  
making it.
I was successful with the little M-ATA from Patton (that's actually an  
OEM) for T38.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 13/05/2010 ? 16:37, Garrison Hoffman a ?crit :

> I'm finally convinced my Grandstream HT386 is too much trouble.  
> There's
> got to be some clever engineering in a device which manages to have a
> problem with every FS configuration I've tested.
>
> What I need now is another dual FXS adapter, one port is for a  
> cordless
> phone and the other is a dedicated fax line.
>
> I'm considering an unlocked Linksys PAP2T, and a Grandstream HT502 but
> I'm open to other recommendations. The HT502 is only on my list due to
> UPnP & TLS support and because my GX2020 has always worked well.
>
> Both products are available at voipsupply.com for under $60,  
> although I
> have no fixed price cutoff.
>
> -gh
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From peder at networkoblivion.com  Thu May 13 08:44:48 2010
From: peder at networkoblivion.com (Peder)
Date: Thu, 13 May 2010 10:44:48 -0500
Subject: [Freeswitch-users] Recommendations for adapter replacement ...
In-Reply-To: <1273761473.1407.26.camel@strontium>
References: <1273761473.1407.26.camel@strontium>
Message-ID: <024801caf2b3$37ed3240$a7c796c0$@com>

SPA2102
http://www.voipsupply.com/spa-2102

Has T.38 support too.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Garrison
Hoffman
Sent: Thursday, May 13, 2010 9:38 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Recommendations for adapter replacement ...

I'm finally convinced my Grandstream HT386 is too much trouble. There's
got to be some clever engineering in a device which manages to have a
problem with every FS configuration I've tested.

What I need now is another dual FXS adapter, one port is for a cordless
phone and the other is a dedicated fax line.

I'm considering an unlocked Linksys PAP2T, and a Grandstream HT502 but
I'm open to other recommendations. The HT502 is only on my list due to
UPnP & TLS support and because my GX2020 has always worked well.

Both products are available at voipsupply.com for under $60, although I
have no fixed price cutoff.

-gh


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



From mike at jerris.com  Thu May 13 09:50:54 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 13 May 2010 12:50:54 -0400
Subject: [Freeswitch-users] Alarm Cleared on channel - Suppressing
In-Reply-To: <AANLkTin4TqPxCPv3_aBywRGtFFncaDGMNfffC_g05i_D@mail.gmail.com>
References: <AANLkTin4TqPxCPv3_aBywRGtFFncaDGMNfffC_g05i_D@mail.gmail.com>
Message-ID: <F24D9A84-C9CE-407B-9F40-55748B8D50B9@jerris.com>

This is different than a reset and shouldn't need to be disabled, I suspect its just a log message you should never see.  

Moy- why are we getting logs of this at all?  Is he really going into alarm state or are we just logging a clear on the periodic update even if it wasn't in alarm state?

Mike

On May 13, 2010, at 8:45 AM, lakshmanan ganapathy wrote:

> Hi all,
> I'm having sangoma A102 card and I'm using freetdm. 
> For every 5 minutes, I get "Alarm Cleared on channel 1:1 to 1:31" and in FreeSwitch console.
> 
> One of my friend told that in Asterisk, he will suppress  this by setting resetinterval=never .
> Can any one please tell when this will occur and is there any way in FreeSwitch to suppress it?? please help!.



From msc at freeswitch.org  Thu May 13 09:55:53 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 13 May 2010 09:55:53 -0700
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
In-Reply-To: <4BEB3E83.3040502@gmail.com>
References: <0b27b9ae92e848c6edfcb0b6b20d255f@mail.gmail.com>
	<AANLkTiloS2j7QKkitjSmhECZXANvXnX8NH3fZiBL1qnm@mail.gmail.com>
	<4BEB2B7D.4080406@gmail.com>
	<AANLkTil9T_6yFvKA8FuoicbvpeZOIkI0XRguUkgO25hL@mail.gmail.com>
	<AANLkTikLpby-J0X6BvvutW7XxpgWylqZM3jU6SrREwBp@mail.gmail.com>
	<4BEB3E83.3040502@gmail.com>
Message-ID: <AANLkTimNvBYVRBcbK3T9z5y80hLcs9BCp4uvyRZIuCWp@mail.gmail.com>

On Wed, May 12, 2010 at 4:49 PM, Phone <phone.bytes at gmail.com> wrote:

> Thanks MC,
>
> Can you point me to any example Perl code to start from?
>
> Look in your FS source tree under libs/esl/perl - there are some examples
of simple ESL connections. Also, look at the other languages in the esl
directory for alternative ideas.

-MC
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From mike at jerris.com  Thu May 13 09:57:11 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 13 May 2010 12:57:11 -0400
Subject: [Freeswitch-users] sqlite in modmanaged
In-Reply-To: <AANLkTim8FGWoVY2j3iZdU4n-8lIi8mcktJz6encL8L6R@mail.gmail.com>
References: <AANLkTinKrCXQpJoQC1DdOcKDAB-0NRmepzLApLO4vmvK@mail.gmail.com>
	<4BEB9C8F.6000206@todandlorna.com>
	<AANLkTim8FGWoVY2j3iZdU4n-8lIi8mcktJz6encL8L6R@mail.gmail.com>
Message-ID: <F023C4BA-7880-4CFB-8E6A-EB46EBC0DDEE@jerris.com>

I will go so far to say that we should not even tell people the embedded db is sqlite and we should explicitly change its format so its not accessible.  If you want to share your db with anything, even other things in process in freeswitch, you should be using another database via odbc.

Mike

On May 13, 2010, at 3:51 AM, babak yakhchali wrote:

> u mean this is not a good idea to get registrations from sqlite?? _______________________________________________



From garrison at codefix.net  Thu May 13 10:32:09 2010
From: garrison at codefix.net (Garrison Hoffman)
Date: Thu, 13 May 2010 13:32:09 -0400
Subject: [Freeswitch-users] Recommendations for adapter replacement ...
In-Reply-To: <1273761473.1407.26.camel@strontium>
References: <1273761473.1407.26.camel@strontium>
Message-ID: <1273771929.1407.124.camel@strontium>

Thank you, Peder and David.

I honestly haven't worried much about the fax, we don't use it much and
LAN load is light, but I just read a bit and T38 relay sounds promising.

I have to say I'm a bit wary of VoIP vendors who don't do on-line
ordering, but Patton's SmartLink 4020 has piqued my interest and I'll
check them out. Naturally I'll be comparing them against Peder's
recommendation of the SPA2102, which has moved to the top of my list (at
least until I can get price info from Patton).

Incidentally, switching from specifying a NAT IP to STUN seems to have
helped (even though it gets the same IP), but I won't consider it a
valid test until it sits overnight. In any case T.38 support is
sufficient reason to upgrade.

-gh

P.S. Bonus points if someone can direct me to something which explains
why moving FS off my local LAN means I have to register each extension
using a different SIP port. I suspect it was doing NAT traversal with
neither STUN nor UPnP but I don't know enough to definitively rule out
my FS config.



From garrison at codefix.net  Thu May 13 10:34:59 2010
From: garrison at codefix.net (Garrison Hoffman)
Date: Thu, 13 May 2010 13:34:59 -0400
Subject: [Freeswitch-users] Recommendations for adapter replacement ...
In-Reply-To: <1273761473.1407.26.camel@strontium>
References: <1273761473.1407.26.camel@strontium>
Message-ID: <1273772099.1407.127.camel@strontium>

D'oh! I just realized VoipSupply sells Patton products. Kindly disregard
my unsightly ignorance.


-gh



From jcasale at activenetwerx.com  Thu May 13 10:55:20 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Thu, 13 May 2010 17:55:20 +0000
Subject: [Freeswitch-users] Recommendations for adapter replacement ...
In-Reply-To: <1273771929.1407.124.camel@strontium>
References: <1273761473.1407.26.camel@strontium>
	<1273771929.1407.124.camel@strontium>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E9A06B10@prato.activenetwerx.local>

>Naturally I'll be comparing them against Peder's
>recommendation of the SPA2102, which has moved to the top of my list (at
>least until I can get price info from Patton).

Buddy, my experience is that device works well for fax, but voice
is bad on it. It echoes like hell and support is what you expect
from a consumer grade product...


From moises.silva at gmail.com  Thu May 13 11:09:27 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Thu, 13 May 2010 14:09:27 -0400
Subject: [Freeswitch-users] Alarm Cleared on channel - Suppressing
In-Reply-To: <F24D9A84-C9CE-407B-9F40-55748B8D50B9@jerris.com>
References: <AANLkTin4TqPxCPv3_aBywRGtFFncaDGMNfffC_g05i_D@mail.gmail.com>
	<F24D9A84-C9CE-407B-9F40-55748B8D50B9@jerris.com>
Message-ID: <AANLkTikuDiW9EACjvLi-R0EtrJqWb1JvKEvEuN1mBHrf@mail.gmail.com>

That message is event-based, so you should not get that msg unless your
wanpipe device is bouncing (disconnected/connected) once in a while.

Ping me on IRC and we can arrange a debug session to find out what is going
on.

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com


On Thu, May 13, 2010 at 12:50 PM, Michael Jerris <mike at jerris.com> wrote:

> This is different than a reset and shouldn't need to be disabled, I suspect
> its just a log message you should never see.
>
> Moy- why are we getting logs of this at all?  Is he really going into alarm
> state or are we just logging a clear on the periodic update even if it
> wasn't in alarm state?
>
> Mike
>
> On May 13, 2010, at 8:45 AM, lakshmanan ganapathy wrote:
>
> > Hi all,
> > I'm having sangoma A102 card and I'm using freetdm.
> > For every 5 minutes, I get "Alarm Cleared on channel 1:1 to 1:31" and in
> FreeSwitch console.
> >
> > One of my friend told that in Asterisk, he will suppress  this by setting
> resetinterval=never .
> > Can any one please tell when this will occur and is there any way in
> FreeSwitch to suppress it?? please help!.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jan.berger at video24.no  Thu May 13 11:18:36 2010
From: jan.berger at video24.no (Jan Berger)
Date: Thu, 13 May 2010 20:18:36 +0200
Subject: [Freeswitch-users] Alarm Cleared on channel - Suppressing
In-Reply-To: <AANLkTikuDiW9EACjvLi-R0EtrJqWb1JvKEvEuN1mBHrf@mail.gmail.com>
References: <AANLkTin4TqPxCPv3_aBywRGtFFncaDGMNfffC_g05i_D@mail.gmail.com>
	<F24D9A84-C9CE-407B-9F40-55748B8D50B9@jerris.com>
	<AANLkTikuDiW9EACjvLi-R0EtrJqWb1JvKEvEuN1mBHrf@mail.gmail.com>
Message-ID: <E4A6AAE8271548CCBC18E662335C5A25@dell9400>

Moises,

 

Do the Sangoma cards provide a hdlc spy on L2 between the driver and the
falc's?

 

jan

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Moises
Silva
Sent: 13. mai 2010 20:09
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Alarm Cleared on channel - Suppressing

 

That message is event-based, so you should not get that msg unless your
wanpipe device is bouncing (disconnected/connected) once in a while.

Ping me on IRC and we can arrange a debug session to find out what is going
on.

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com



On Thu, May 13, 2010 at 12:50 PM, Michael Jerris <mike at jerris.com> wrote:

This is different than a reset and shouldn't need to be disabled, I suspect
its just a log message you should never see.

Moy- why are we getting logs of this at all?  Is he really going into alarm
state or are we just logging a clear on the periodic update even if it
wasn't in alarm state?

Mike


On May 13, 2010, at 8:45 AM, lakshmanan ganapathy wrote:

> Hi all,
> I'm having sangoma A102 card and I'm using freetdm.
> For every 5 minutes, I get "Alarm Cleared on channel 1:1 to 1:31" and in
FreeSwitch console.
>
> One of my friend told that in Asterisk, he will suppress  this by setting
resetinterval=never .
> Can any one please tell when this will occur and is there any way in
FreeSwitch to suppress it?? please help!.



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

 

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From info at evestech.com  Thu May 13 11:21:59 2010
From: info at evestech.com (Kashif Kahn)
Date: Thu, 13 May 2010 11:21:59 -0700 (PDT)
Subject: [Freeswitch-users] Robust Affordable Speech Recognition
In-Reply-To: <FCC1A29F6BF74D48AB906CFC2BC18CA2@dell9400>
References: <600175.94339.qm@web208.biz.mail.re2.yahoo.com>
	<FCC1A29F6BF74D48AB906CFC2BC18CA2@dell9400>
Message-ID: <598954.24948.qm@web207.biz.mail.re2.yahoo.com>

Hi Jan,

Our answers to your questions are as 
follows. Please note that there is a wide variety of information 
available on our website under "Speech Engine" pull-down menu. You 
should also review Freeswitch section under "Soft-PBX" for free connector and engine pricing information. Our website is: http://www.vestec.ca/

1) What languages 
do you
support

We currently support American English and 
working on major European and Asian language acoustic models as well.

2) What are your 
recognition
stats per language?

Three points are worthy of note here: 

(a) we have rigorously benchmarked our recognition 
accuracy against leading commercial engines in the market and can safely say that we deliver among the highest recognition accuracy in the 
industry; 

(b) generally speaking, a native speaker can expect a 
recognition accuracy - without "tuning" - in the 90% range while a 
non-native speaker can expect a recognition accuracy - again, without 
"tuning" - in the 80% range. Grammar "tuning" - for example, via 
addition of custom pronunciations for difficult to recognize words - 
generally improves recognition accuracy by an additional 5-10%; 

(c) in comparative testing against some of the leading commercial speech 
engines, we scored an accuracy improvement of over 3%. See: http://www.vestec.ca/recog_acc

3) What are the benefits 
of
using this compared to Sphinx/Pocketsphinx?

There 
are two fundamental advantages of using Vestec over Sphinx: 

(a) 
Vestec speech engine comes with a robust acoustic model while for Sphinx you need to develop your own acoustic model. (Acoustic model is the 
"ear drum" if you will that does the recognition). Now, development of a high quality acoustic model is no mean feat, even if you have the 
necessary tools. For starters you need high quality training data that 
can cost tens of thousands of dollars. Next, you need highly specialized knowledge in order to properly manipulate the various parameters of 
training tools for optimal recognition quality. Finally, you need to be 
fully prepared to undertake a trial-and-error process - sometimes 
spanning 6-8 months - to achieve the desired recognition accuracy 
results, with the right data and training tools. 

(b) Unlike 
Sphinx, Vestec speech engine supports industry standard grammar writing format. This allows you to port your existing standard 
grammars to the engine as well as make sure that your work is reusable. 
Similarly, we provide a variety of useful grammar writing tools and 
functions that are not available with Sphinx to make the developer's job a lot easier.

4) How does the (1)
installation of a license happen, (2) license check happen.

You need to put the license files that you purchase from Vestec webstore
under a specific directory. The engine then takes care of the rest. The 
license check is host based. In other words, every license is
bounded to one physical/virtual machine. You cannot buy one port license and
expect to run the speech engine on several machines.

 
5) Does the licensing server
support redundancy schemes? 
We used to have a 
redundancy scheme but recently disabled it in preparation for the 
release of a new architecture. The redundancy scheme will be integrated 
in the new architecture that will support MRCP. Please note that 
software maintenance - covering patches and major upgrades - is free for the first 12 months since date of license purchase. So, you will be 
able to transition to the new architecture over next several months if 
you were to purchase a license now.

Hope this helps.

Best 
regards,
-Kashif




________________________________
From: Jan Berger <jan.berger at video24.no>
To: freeswitch-users at lists.freeswitch.org
Sent: Thu, May 13, 2010 11:36:54 AM
Subject: Re: [Freeswitch-users] Robust Affordable Speech Recognition

 
Hi,
 
Hope you don?t mind
a few nosy questions.
 
What languages do you
support (1) national characters, (2) pre-build databases.
 
What are your recognition
stats per language?
 
What are the benefits of
using this compared to Sphinx/Pocketsphinx?
 
How does the (1)
installation of a license happen, (2) license check happen.
 
Does the licensing server
support redundancy schemes? 
 
Jan
 

________________________________
 
From:freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kashif Kahn
Sent: 13. mai 2010 16:04
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Robust
Affordable Speech Recognition
 
Dear All,

All those who have wanted a speech recognition solution for Freeswitch but
found the software cost too expensive or the recognition accuracy
unsatisfactory, please consider Vestec Speech Engine for Freeswitch at: http://www.vestec.ca/products A starter kit - which is a specially priced one port (ie. one channel) license
for the standard engine - is available for only $25. Additional ports
(channels) licenses can be purchased for $99/port. The engine comes with a
free-of-charge Freeswitch connector, thereby allowing direct interaction via
Dialplan.

Best regards,
-Kashif
 
Kashif Kahn
VP, Business Development
Vestec, Inc.
Waterloo, ON Canada
phone: (519) 885-7615
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From david.ponzone at gmail.com  Thu May 13 11:42:13 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Thu, 13 May 2010 20:42:13 +0200
Subject: [Freeswitch-users] Recommendations for adapter replacement ...
In-Reply-To: <1273771929.1407.124.camel@strontium>
References: <1273761473.1407.26.camel@strontium>
	<1273771929.1407.124.camel@strontium>
Message-ID: <0ECBA35F-00F6-4D0F-B7A7-427DE65C6B9E@gmail.com>

Garrison,

just to be sure: does your device set rport in the INVITE ?
If not, it could explain your issue, and you could force that by using  
force-rport in the SIP profile.

Bonus: I dont think you need to do that. If your FS is out of your  
local LAN, and using public IP, you can register as many devices on  
your local LAN without doing anything special.
The only thing is that FS does some magic to guess that a device is  
behind NAT (based on the IPs in the INVITE, on the presence of rport  
in the Via field, etc...), and sometimes, the magic is not powerful  
enough (if rport is not there for example).
Using force-rport can help.
To check if FS sees a device correctly behind NAT, check the  
sofia_contact (sofia status profile <profile name> shows that for all  
registered devices).
You should see fs_nat=yes and fs_path should contain the right way to  
reach the device (with public Ip and public port).

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 13/05/2010 ? 19:32, Garrison Hoffman a ?crit :

> Thank you, Peder and David.
>
> I honestly haven't worried much about the fax, we don't use it much  
> and
> LAN load is light, but I just read a bit and T38 relay sounds  
> promising.
>
> I have to say I'm a bit wary of VoIP vendors who don't do on-line
> ordering, but Patton's SmartLink 4020 has piqued my interest and I'll
> check them out. Naturally I'll be comparing them against Peder's
> recommendation of the SPA2102, which has moved to the top of my list  
> (at
> least until I can get price info from Patton).
>
> Incidentally, switching from specifying a NAT IP to STUN seems to have
> helped (even though it gets the same IP), but I won't consider it a
> valid test until it sits overnight. In any case T.38 support is
> sufficient reason to upgrade.
>
> -gh
>
> P.S. Bonus points if someone can direct me to something which explains
> why moving FS off my local LAN means I have to register each extension
> using a different SIP port. I suspect it was doing NAT traversal with
> neither STUN nor UPnP but I don't know enough to definitively rule out
> my FS config.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From garrison at codefix.net  Thu May 13 11:52:16 2010
From: garrison at codefix.net (Garrison Hoffman)
Date: Thu, 13 May 2010 14:52:16 -0400
Subject: [Freeswitch-users] Recommendations for adapter replacement ...
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E9A06B10@prato.activenetwerx.local>
References: <1273761473.1407.26.camel@strontium>
	<1273771929.1407.124.camel@strontium>
	<CA5A491E9DEFBE4CB777DE97E21575E9A06B10@prato.activenetwerx.local>
Message-ID: <1273776736.1407.173.camel@strontium>

On Thu, 2010-05-13 at 17:55 +0000, Joseph L. Casale wrote:
> >Naturally I'll be comparing them against Peder's
> >recommendation of the SPA2102, which has moved to the top of my list (at
> >least until I can get price info from Patton).
> 
> Buddy, my experience is that device works well for fax, but voice
> is bad on it. It echoes like hell and support is what you expect
> from a consumer grade product...

Well, it's a dynamic list :-D

I have enough experience with Linksys/Cisco routers that I had some
hesitation even as I wrote that, meanwhile I've discovered that my
preferred vendor apparently sells Patton and I'm seriously considering
http://www.voipsupply.com/sl4022 T.38 isn't listed there but is on
Patton's site.

There's an obvious bias on this list against, as you say, consumer grade
products, which often work fine within limited parameters, but I'm
finding that when it comes to VoIP hardware those limitations are more
frequently exceeding my tolerance. In other words, a 100% difference in
price counts for less than a 50% difference in the amount of time I need
to spend getting it to work. 

-gh



From pjintheusa at gmail.com  Thu May 13 12:01:16 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Thu, 13 May 2010 15:01:16 -0400
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
In-Reply-To: <AANLkTimNvBYVRBcbK3T9z5y80hLcs9BCp4uvyRZIuCWp@mail.gmail.com>
References: <0b27b9ae92e848c6edfcb0b6b20d255f@mail.gmail.com>
	<AANLkTiloS2j7QKkitjSmhECZXANvXnX8NH3fZiBL1qnm@mail.gmail.com>
	<4BEB2B7D.4080406@gmail.com>
	<AANLkTil9T_6yFvKA8FuoicbvpeZOIkI0XRguUkgO25hL@mail.gmail.com>
	<AANLkTikLpby-J0X6BvvutW7XxpgWylqZM3jU6SrREwBp@mail.gmail.com>
	<4BEB3E83.3040502@gmail.com>
	<AANLkTimNvBYVRBcbK3T9z5y80hLcs9BCp4uvyRZIuCWp@mail.gmail.com>
Message-ID: <AANLkTik0JtsIKBLs-6ZJ2PxVq-esBBJe4dGiAiNLKhyG@mail.gmail.com>

And just a general thought of using ESL vs an in process solution like
mod_managed, or LUA.

My understanding is, that using a separate server/process does potentially
give you another point of failure and, if you use a single ESL server
application to control several FS boxes, potentially a single point of
failure. It is fairly easy to build a scalable and reliable FS cluster and
using DNS SRV and OpenSIPS in order to avoid any single points of failure.
Having independent FS boxes that pull data, but can fail with little impact
seems attractive to me,

On Thu, May 13, 2010 at 12:55 PM, Michael Collins <msc at freeswitch.org>wrote:

>
>
> On Wed, May 12, 2010 at 4:49 PM, Phone <phone.bytes at gmail.com> wrote:
>
>> Thanks MC,
>>
>> Can you point me to any example Perl code to start from?
>>
>> Look in your FS source tree under libs/esl/perl - there are some examples
> of simple ESL connections. Also, look at the other languages in the esl
> directory for alternative ideas.
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kris at kriskinc.com  Thu May 13 12:15:37 2010
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Thu, 13 May 2010 15:15:37 -0400
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
In-Reply-To: <AANLkTik0JtsIKBLs-6ZJ2PxVq-esBBJe4dGiAiNLKhyG@mail.gmail.com>
References: <0b27b9ae92e848c6edfcb0b6b20d255f@mail.gmail.com>
	<AANLkTiloS2j7QKkitjSmhECZXANvXnX8NH3fZiBL1qnm@mail.gmail.com>
	<4BEB2B7D.4080406@gmail.com>
	<AANLkTil9T_6yFvKA8FuoicbvpeZOIkI0XRguUkgO25hL@mail.gmail.com>
	<AANLkTikLpby-J0X6BvvutW7XxpgWylqZM3jU6SrREwBp@mail.gmail.com>
	<4BEB3E83.3040502@gmail.com>
	<AANLkTimNvBYVRBcbK3T9z5y80hLcs9BCp4uvyRZIuCWp@mail.gmail.com>
	<AANLkTik0JtsIKBLs-6ZJ2PxVq-esBBJe4dGiAiNLKhyG@mail.gmail.com>
Message-ID: <AANLkTin_GcxWwk7otWFDzlA95XnoYSiVgvyky1mHLzGG@mail.gmail.com>

ESL connecting to a socket actually proves to be more scalable and
more available with fewer components.  Why not have N servers running
your socket app with N servers running FreeSWITCH?

OpenSIPS introduces its own issues with failover and I've yet to see
DNS SRV be the reliability/scalability solution it's made out to be.

On Thu, May 13, 2010 at 3:01 PM, Phillip Jones <pjintheusa at gmail.com> wrote:
> And just a general thought of using ESL vs an in process solution like
> mod_managed, or LUA.
>
> My understanding is, that using a separate server/process does potentially
> give you another point of failure and, if you use a single ESL server
> application to control several FS boxes, potentially a single point of
> failure. It is fairly easy to build a scalable and reliable FS cluster and
> using DNS SRV and OpenSIPS in order to avoid any single points of failure.
> Having independent FS boxes that pull data, but can fail with little impact
> seems attractive to me,

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From anthony.minessale at gmail.com  Thu May 13 12:16:46 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 13 May 2010 14:16:46 -0500
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
In-Reply-To: <AANLkTik0JtsIKBLs-6ZJ2PxVq-esBBJe4dGiAiNLKhyG@mail.gmail.com>
References: <0b27b9ae92e848c6edfcb0b6b20d255f@mail.gmail.com>
	<AANLkTiloS2j7QKkitjSmhECZXANvXnX8NH3fZiBL1qnm@mail.gmail.com>
	<4BEB2B7D.4080406@gmail.com>
	<AANLkTil9T_6yFvKA8FuoicbvpeZOIkI0XRguUkgO25hL@mail.gmail.com>
	<AANLkTikLpby-J0X6BvvutW7XxpgWylqZM3jU6SrREwBp@mail.gmail.com>
	<4BEB3E83.3040502@gmail.com>
	<AANLkTimNvBYVRBcbK3T9z5y80hLcs9BCp4uvyRZIuCWp@mail.gmail.com>
	<AANLkTik0JtsIKBLs-6ZJ2PxVq-esBBJe4dGiAiNLKhyG@mail.gmail.com>
Message-ID: <AANLkTilKOUlc9NjE2Lfyz3qWJHKwHm9w8ocwzYEO92fU@mail.gmail.com>

Don't forget C, a loadable C module is the most robust way to make an
application.
You can write even ESL apps in C too.




On Thu, May 13, 2010 at 2:01 PM, Phillip Jones <pjintheusa at gmail.com> wrote:

> And just a general thought of using ESL vs an in process solution like
> mod_managed, or LUA.
>
> My understanding is, that using a separate server/process does potentially
> give you another point of failure and, if you use a single ESL server
> application to control several FS boxes, potentially a single point of
> failure. It is fairly easy to build a scalable and reliable FS cluster and
> using DNS SRV and OpenSIPS in order to avoid any single points of failure.
> Having independent FS boxes that pull data, but can fail with little impact
> seems attractive to me,
>
> On Thu, May 13, 2010 at 12:55 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Wed, May 12, 2010 at 4:49 PM, Phone <phone.bytes at gmail.com> wrote:
>>
>>> Thanks MC,
>>>
>>> Can you point me to any example Perl code to start from?
>>>
>>> Look in your FS source tree under libs/esl/perl - there are some examples
>> of simple ESL connections. Also, look at the other languages in the esl
>> directory for alternative ideas.
>>
>> -MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <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
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From jan.berger at video24.no  Thu May 13 12:40:31 2010
From: jan.berger at video24.no (Jan Berger)
Date: Thu, 13 May 2010 21:40:31 +0200
Subject: [Freeswitch-users] Robust Affordable Speech Recognition
In-Reply-To: <598954.24948.qm@web207.biz.mail.re2.yahoo.com>
References: <600175.94339.qm@web208.biz.mail.re2.yahoo.com>
	<FCC1A29F6BF74D48AB906CFC2BC18CA2@dell9400>
	<598954.24948.qm@web207.biz.mail.re2.yahoo.com>
Message-ID: <A47DB7C423864372B68C992B951D0EF6@dell9400>

Sadly that rules me out as I can't do much with US English. 

 

Jan

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kashif
Kahn
Sent: 13. mai 2010 20:22
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Robust Affordable Speech Recognition

 

Hi Jan,

Our answers to your questions are as follows. Please note that there is a
wide variety of information available on our website under "Speech Engine"
pull-down menu. You should also review Freeswitch section under "Soft-PBX"
for free connector and engine pricing information. Our website is:
http://www.vestec.ca/

 

1) What languages do you support

We currently support American English and working on major European and
Asian language acoustic models as well.

2) What are your recognition stats per language?

Three points are worthy of note here: 

(a) we have rigorously benchmarked our recognition accuracy against leading
commercial engines in the market and can safely say that we deliver among
the highest recognition accuracy in the industry; 

(b) generally speaking, a native speaker can expect a recognition accuracy -
without "tuning" - in the 90% range while a non-native speaker can expect a
recognition accuracy - again, without "tuning" - in the 80% range. Grammar
"tuning" - for example, via addition of custom pronunciations for difficult
to recognize words - generally improves recognition accuracy by an
additional 5-10%; 

(c) in comparative testing against some of the leading commercial speech
engines, we scored an accuracy improvement of over 3%. See:
http://www.vestec.ca/recog_acc

3) What are the benefits of using this compared to Sphinx/Pocketsphinx?

There are two fundamental advantages of using Vestec over Sphinx: 

(a) Vestec speech engine comes with a robust acoustic model while for Sphinx
you need to develop your own acoustic model. (Acoustic model is the "ear
drum" if you will that does the recognition). Now, development of a high
quality acoustic model is no mean feat, even if you have the necessary
tools. For starters you need high quality training data that can cost tens
of thousands of dollars. Next, you need highly specialized knowledge in
order to properly manipulate the various parameters of training tools for
optimal recognition quality. Finally, you need to be fully prepared to
undertake a trial-and-error process - sometimes spanning 6-8 months - to
achieve the desired recognition accuracy results, with the right data and
training tools. 

(b) Unlike Sphinx, Vestec speech engine supports industry standard grammar
writing format. This allows you to port your existing standard grammars to
the engine as well as make sure that your work is reusable. Similarly, we
provide a variety of useful grammar writing tools and functions that are not
available with Sphinx to make the developer's job a lot easier.

4) How does the (1) installation of a license happen, (2) license check
happen.

You need to put the license files that you purchase from Vestec webstore
under a specific directory. The engine then takes care of the rest. The
license check is host based. In other words, every license is bounded to one
physical/virtual machine. You cannot buy one port license and expect to run
the speech engine on several machines.

5) Does the licensing server support redundancy schemes? 


We used to have a redundancy scheme but recently disabled it in preparation
for the release of a new architecture. The redundancy scheme will be
integrated in the new architecture that will support MRCP. Please note that
software maintenance - covering patches and major upgrades - is free for the
first 12 months since date of license purchase. So, you will be able to
transition to the new architecture over next several months if you were to
purchase a license now.

Hope this helps.

Best regards,
-Kashif

 

 

  _____  

From: Jan Berger <jan.berger at video24.no>
To: freeswitch-users at lists.freeswitch.org
Sent: Thu, May 13, 2010 11:36:54 AM
Subject: Re: [Freeswitch-users] Robust Affordable Speech Recognition

Hi,

 

Hope you don't mind a few nosy questions.

 

What languages do you support (1) national characters, (2) pre-build
databases.

 

What are your recognition stats per language?

 

What are the benefits of using this compared to Sphinx/Pocketsphinx?

 

How does the (1) installation of a license happen, (2) license check happen.

 

Does the licensing server support redundancy schemes? 

 

Jan

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kashif
Kahn
Sent: 13. mai 2010 16:04
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Robust Affordable Speech Recognition

 

Dear All,

All those who have wanted a speech recognition solution for Freeswitch but
found the software cost too expensive or the recognition accuracy
unsatisfactory, please consider Vestec Speech Engine for Freeswitch at:
http://www.vestec.ca/products A starter kit - which is a specially priced
one port (ie. one channel) license for the standard engine - is available
for only $25. Additional ports (channels) licenses can be purchased for
$99/port. The engine comes with a free-of-charge Freeswitch connector,
thereby allowing direct interaction via Dialplan.

Best regards,
-Kashif

 

Kashif Kahn
VP, Business Development
Vestec, Inc.
Waterloo, ON Canada
phone: (519) 885-7615

 

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From moises.silva at gmail.com  Thu May 13 12:43:29 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Thu, 13 May 2010 15:43:29 -0400
Subject: [Freeswitch-users] Alarm Cleared on channel - Suppressing
In-Reply-To: <E4A6AAE8271548CCBC18E662335C5A25@dell9400>
References: <AANLkTin4TqPxCPv3_aBywRGtFFncaDGMNfffC_g05i_D@mail.gmail.com>
	<F24D9A84-C9CE-407B-9F40-55748B8D50B9@jerris.com>
	<AANLkTikuDiW9EACjvLi-R0EtrJqWb1JvKEvEuN1mBHrf@mail.gmail.com>
	<E4A6AAE8271548CCBC18E662335C5A25@dell9400>
Message-ID: <AANLkTil1kPWAmoGMSSjRvpdm733iWACSHPOFAceRYPOb@mail.gmail.com>

Let's try to not hijack threads :-)

I was not familiar with the term "FALC", after googling it I must assume you
are talking about the framer ( falc seems to be an infineon framer? ),
correct me if I'm wrong.

The functionality you look for is probably achieved through wanpipemon -i
<interface> -pcap -pcap_file isdn.pcap -prot ISDN -full -systime -c trd

That will trace the D-channel frames and dump them in a pcap file.

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com


On Thu, May 13, 2010 at 2:18 PM, Jan Berger <jan.berger at video24.no> wrote:

>  Moises,
>
>
>
> Do the Sangoma cards provide a hdlc spy on L2 between the driver and the
> falc?s?
>
>
>
> jan
>
>
>  ------------------------------
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Moises Silva
> *Sent:* 13. mai 2010 20:09
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Alarm Cleared on channel - Suppressing
>
>
>
> That message is event-based, so you should not get that msg unless your
> wanpipe device is bouncing (disconnected/connected) once in a while.
>
> Ping me on IRC and we can arrange a debug session to find out what is going
> on.
>
> Moises Silva
> Senior Software Engineer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
> 9T3 Canada
> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>
>  On Thu, May 13, 2010 at 12:50 PM, Michael Jerris <mike at jerris.com> wrote:
>
> This is different than a reset and shouldn't need to be disabled, I suspect
> its just a log message you should never see.
>
> Moy- why are we getting logs of this at all?  Is he really going into alarm
> state or are we just logging a clear on the periodic update even if it
> wasn't in alarm state?
>
> Mike
>
>
> On May 13, 2010, at 8:45 AM, lakshmanan ganapathy wrote:
>
> > Hi all,
> > I'm having sangoma A102 card and I'm using freetdm.
> > For every 5 minutes, I get "Alarm Cleared on channel 1:1 to 1:31" and in
> FreeSwitch console.
> >
> > One of my friend told that in Asterisk, he will suppress  this by setting
> resetinterval=never .
> > Can any one please tell when this will occur and is there any way in
> FreeSwitch to suppress it?? please help!.
>
>    _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From pjintheusa at gmail.com  Thu May 13 13:16:04 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Thu, 13 May 2010 16:16:04 -0400
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
In-Reply-To: <AANLkTin_GcxWwk7otWFDzlA95XnoYSiVgvyky1mHLzGG@mail.gmail.com>
References: <0b27b9ae92e848c6edfcb0b6b20d255f@mail.gmail.com>
	<AANLkTiloS2j7QKkitjSmhECZXANvXnX8NH3fZiBL1qnm@mail.gmail.com>
	<4BEB2B7D.4080406@gmail.com>
	<AANLkTil9T_6yFvKA8FuoicbvpeZOIkI0XRguUkgO25hL@mail.gmail.com>
	<AANLkTikLpby-J0X6BvvutW7XxpgWylqZM3jU6SrREwBp@mail.gmail.com>
	<4BEB3E83.3040502@gmail.com>
	<AANLkTimNvBYVRBcbK3T9z5y80hLcs9BCp4uvyRZIuCWp@mail.gmail.com>
	<AANLkTik0JtsIKBLs-6ZJ2PxVq-esBBJe4dGiAiNLKhyG@mail.gmail.com>
	<AANLkTin_GcxWwk7otWFDzlA95XnoYSiVgvyky1mHLzGG@mail.gmail.com>
Message-ID: <AANLkTikTRItehHsdKYi6NUHbIWKrHtaTlT6fJQ8MHV2J@mail.gmail.com>

What would the advantage of an N to N architecture be though? An ESL server
controlling several FS instances has a view of everything that is going on.
All calls/conferences etc. A real advantage. That is lost when two ESL
servers are run in parallel. State information could be in a DB cluster -
but why not have the in-process app access this directly, cutting out the
middle man?

How does using ESL make it more scalable and more available with fewer
components? I am sure you are correct - I just don't see it.

On Thu, May 13, 2010 at 3:15 PM, Kristian Kielhofner <kris at kriskinc.com>wrote:

> ESL connecting to a socket actually proves to be more scalable and
> more available with fewer components.  Why not have N servers running
> your socket app with N servers running FreeSWITCH?
>
> OpenSIPS introduces its own issues with failover and I've yet to see
> DNS SRV be the reliability/scalability solution it's made out to be.
>
> On Thu, May 13, 2010 at 3:01 PM, Phillip Jones <pjintheusa at gmail.com>
> wrote:
> > And just a general thought of using ESL vs an in process solution like
> > mod_managed, or LUA.
> >
> > My understanding is, that using a separate server/process does
> potentially
> > give you another point of failure and, if you use a single ESL server
> > application to control several FS boxes, potentially a single point of
> > failure. It is fairly easy to build a scalable and reliable FS cluster
> and
> > using DNS SRV and OpenSIPS in order to avoid any single points of
> failure.
> > Having independent FS boxes that pull data, but can fail with little
> impact
> > seems attractive to me,
>
> --
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Thu May 13 13:28:00 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 13 May 2010 13:28:00 -0700
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
In-Reply-To: <AANLkTikTRItehHsdKYi6NUHbIWKrHtaTlT6fJQ8MHV2J@mail.gmail.com>
References: <0b27b9ae92e848c6edfcb0b6b20d255f@mail.gmail.com>
	<AANLkTiloS2j7QKkitjSmhECZXANvXnX8NH3fZiBL1qnm@mail.gmail.com>
	<4BEB2B7D.4080406@gmail.com>
	<AANLkTil9T_6yFvKA8FuoicbvpeZOIkI0XRguUkgO25hL@mail.gmail.com>
	<AANLkTikLpby-J0X6BvvutW7XxpgWylqZM3jU6SrREwBp@mail.gmail.com>
	<4BEB3E83.3040502@gmail.com>
	<AANLkTimNvBYVRBcbK3T9z5y80hLcs9BCp4uvyRZIuCWp@mail.gmail.com>
	<AANLkTik0JtsIKBLs-6ZJ2PxVq-esBBJe4dGiAiNLKhyG@mail.gmail.com>
	<AANLkTin_GcxWwk7otWFDzlA95XnoYSiVgvyky1mHLzGG@mail.gmail.com>
	<AANLkTikTRItehHsdKYi6NUHbIWKrHtaTlT6fJQ8MHV2J@mail.gmail.com>
Message-ID: <AANLkTim_-_JGeTba2jWr5hxfakt7ymQVFdKuqzWTP4PG@mail.gmail.com>

Guys, this is getting a bit more intense than what the OP asked about. He
wanted some information on how to use a first-aid kit, not how to perform
brain surgery! :P

-MC

On Thu, May 13, 2010 at 1:16 PM, Phillip Jones <pjintheusa at gmail.com> wrote:

> What would the advantage of an N to N architecture be though? An ESL server
> controlling several FS instances has a view of everything that is going on.
> All calls/conferences etc. A real advantage. That is lost when two ESL
> servers are run in parallel. State information could be in a DB cluster -
> but why not have the in-process app access this directly, cutting out the
> middle man?
>
> How does using ESL make it more scalable and more available with fewer
> components? I am sure you are correct - I just don't see it.
>
> On Thu, May 13, 2010 at 3:15 PM, Kristian Kielhofner <kris at kriskinc.com>wrote:
>
>> ESL connecting to a socket actually proves to be more scalable and
>> more available with fewer components.  Why not have N servers running
>> your socket app with N servers running FreeSWITCH?
>>
>> OpenSIPS introduces its own issues with failover and I've yet to see
>> DNS SRV be the reliability/scalability solution it's made out to be.
>>
>> On Thu, May 13, 2010 at 3:01 PM, Phillip Jones <pjintheusa at gmail.com>
>> wrote:
>> > And just a general thought of using ESL vs an in process solution like
>> > mod_managed, or LUA.
>> >
>> > My understanding is, that using a separate server/process does
>> potentially
>> > give you another point of failure and, if you use a single ESL server
>> > application to control several FS boxes, potentially a single point of
>> > failure. It is fairly easy to build a scalable and reliable FS cluster
>> and
>> > using DNS SRV and OpenSIPS in order to avoid any single points of
>> failure.
>> > Having independent FS boxes that pull data, but can fail with little
>> impact
>> > seems attractive to me,
>>
>> --
>> Kristian Kielhofner
>> http://www.astlinux.org
>> http://blog.krisk.org
>> http://www.star2star.com
>> http://www.submityoursip.com
>> http://www.voalte.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bwibowo at gmail.com  Thu May 13 14:49:39 2010
From: bwibowo at gmail.com (budi wibowo)
Date: Fri, 14 May 2010 04:49:39 +0700
Subject: [Freeswitch-users] mod_radius_cdr invalid ELF header
Message-ID: <AANLkTikcM7B-SzB7t89dyMQp6Ivhbbo3OgHTafyDYESg@mail.gmail.com>

dear list
i follow http://wiki.freeswitch.org/wiki/Mod_radius_cdr, but onexecution i
found some error

2010-05-14 04:42:00.182283 [CRIT] switch_loadable_module.c:871 Error Loading
module /usr/local/freeswitch/mod/mod_radius_cdr.so
**/usr/local/freeswitch/mod/mod_radius_cdr.so: invalid ELF header**

trying to google but luck

regards

budi wibowo
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From brian at freeswitch.org  Thu May 13 15:01:26 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 13 May 2010 17:01:26 -0500
Subject: [Freeswitch-users] mod_radius_cdr invalid ELF header
In-Reply-To: <AANLkTikcM7B-SzB7t89dyMQp6Ivhbbo3OgHTafyDYESg@mail.gmail.com>
References: <AANLkTikcM7B-SzB7t89dyMQp6Ivhbbo3OgHTafyDYESg@mail.gmail.com>
Message-ID: <62B274BC-8F33-48E3-9BC6-6D135ED773A8@freeswitch.org>

I'm going to guess that you copied the .so file in the dir and tried to load it... but thanks to the wonders of libtool you actually copied a bash script the real mod_radius_cdr.so is in .libs/mod_radius_cdr.so this is why you type make mod_radius_cdr-install and you won't encounter this.

/b

On May 13, 2010, at 4:49 PM, budi wibowo wrote:

> dear list
> i follow http://wiki.freeswitch.org/wiki/Mod_radius_cdr, but onexecution i found some error 
> 
> 2010-05-14 04:42:00.182283 [CRIT] switch_loadable_module.c:871 Error Loading module /usr/local/freeswitch/mod/mod_radius_cdr.so
> **/usr/local/freeswitch/mod/mod_radius_cdr.so: invalid ELF header**
> 
> trying to google but luck
> 
> regards
> 
> budi wibowo

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From neilp at cs.stanford.edu  Thu May 13 17:14:22 2010
From: neilp at cs.stanford.edu (Neil Patel)
Date: Thu, 13 May 2010 17:14:22 -0700
Subject: [Freeswitch-users] how do I detect a call has rung once?
In-Reply-To: <AANLkTinnTRKaOUI2OHSoi9Lu1jjbSfTxZjFBn_9EIMa1@mail.gmail.com>
References: <u2je5503bdb1005101403u405a5eebo9a2b8c698e9cffaf@mail.gmail.com> 
	<201005101734.09903.sos@sokhapkin.dyndns.org>
	<AANLkTilVHAdZDgF3a5Mg2HgXrXOg6hkrLKGb9xUAAdik@mail.gmail.com> 
	<AANLkTikqOpUTWkv9izTLM8CKljgexflNPPEVUxiYzw01@mail.gmail.com> 
	<E08F3AA9-613E-446A-BD22-5B31F38EB78C@freeswitch.org>
	<AANLkTimHH6zzA_-Dal1YkHlSL4gkbp7m8TyoFrukBmB3@mail.gmail.com> 
	<AANLkTinHQnCTPTDtxXRl80Pd9aGMs8V2Xws1z5KeMegU@mail.gmail.com> 
	<AANLkTilcSOp1nfZxSqtndA_HBvKNl1iPrOwJk4uC3P4z@mail.gmail.com> 
	<AANLkTiklAqw6Er_0fCq9CQnt6RyEPY9P0Usg5iAjl4nw@mail.gmail.com> 
	<AANLkTinnTRKaOUI2OHSoi9Lu1jjbSfTxZjFBn_9EIMa1@mail.gmail.com>
Message-ID: <AANLkTilt2WwQJmSoOqraooUdREErvKWL_sxi6ZlKiPUY@mail.gmail.com>

Thanks Michael, I was looking for code examples to receive events over ESL,
and found them in libs/esl/.

I'm still not clear on the following:

   1. I have a controller script that originates calls; it runs "show
   channels" between calls and parses out the number of active channels. If
   below threshold, originate next call. If not, wait a couple seconds and run
   "show channels" again, repeating till below threshold. Is this the best way?
   Can this be done more efficiently through events? How?
   2. For placing missed calls to SIP endpoints, it sounds like* *I can wait
   for a signal 180 or 183 to know when to hang up. How do I check for those
   signals? Is this another job for events?
   3. For placing missed calls over openzap/PRI, it sounds like I can check
   for call states PROGRESS or ALERTING? Where are those states from, and how
   do I check them? Again, can I use events? The other option you gave is
   checking for endpoint_disposition. This is in the cdr, but how do I use it
   for detecting when I should hang up for a missed call? Is there an event
   associated with it? What are the call states it goes through?

-Neil

On Wed, May 12, 2010 at 2:47 PM, Michael Collins <msc at freeswitch.org> wrote:

>
> On Wed, May 12, 2010 at 11:59 AM, Neil Patel <neilp at cs.stanford.edu>wrote:
>
>> Thanks for the tips.
>>
>> Where are there examples of using ESL to register and handle events? For
>> e.g. I don't see any example of how to register for a change in channel
>> variable endpoint_disposition. Or even to check the CHANNEL_STATE event.
>>
>> -Neil
>>
>> Neil,
>
> Check out the event socket documentation, specifically the "events plain
> all" and "filter" commands. You can listen to all events or just specific
> ones:
> http://wiki.freeswitch.org/wiki/Event_socket
>
> If you want to experiment with the event socket and the various events then
> use fs_cli. Connect with fs_cli and then issue "/log 0" which turns off all
> debug messages. Now you have basically a raw event socket. Try some of these
> commands:
>
> /event plain all
>
> watch the fun messages fly. :)
>
> You can filter them as well:
>
> /filter Event-Name CHANNEL_STATE
> /filter Event-Name CHANNEL_HANGUP
>
> when you apply a filter you will receive only those things you choose. A
> it's a "filter in" not filter out. Another way of saying it is "show me
> events named CHANNEL_STATE" and "show me events named CHANNEL_HANGUP"
>
> just note that in your script you will be using the ESL abstractions for
> these:
>
> http://wiki.freeswitch.org/wiki/Esl#ESLconnection_Object
>
> I hope that helps! Have fun.
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From freeswitch at gilligan.id.au  Thu May 13 18:10:48 2010
From: freeswitch at gilligan.id.au (Chris)
Date: Fri, 14 May 2010 11:10:48 +1000
Subject: [Freeswitch-users] mod_managed or a script,
	sending DTMF and 	receiving DTMF
In-Reply-To: <AANLkTik1BYHsuHlfS9FAD3pEGS5GaEIINiiY9708sFji@mail.gmail.com>
References: <AANLkTikSTwOw1yiBGO9VrVDnsKtH7ySz8V2LxL1XUdru@mail.gmail.com> 
	<AANLkTik1BYHsuHlfS9FAD3pEGS5GaEIINiiY9708sFji@mail.gmail.com>
Message-ID: <AANLkTimDZx2T15GHnWck36bESESuK8n4u_wzzujyxvfE@mail.gmail.com>

Thanks so that is the only way?  I will have to do some tests and see if i
can capture dtmf at the same time as i send when i do it that way.  Any
other options?

On Thu, May 13, 2010 at 10:50 PM, Phillip Jones <pjintheusa at gmail.com>wrote:

> Try using http://wiki.freeswitch.org/wiki/Mod_commands#uuid_send_dtmf
>
> On Thu, May 13, 2010 at 5:32 AM, Chris <freeswitch at gilligan.id.au> wrote:
>
>> HI,
>> I am currently working on a project in mod_managed and i am trying to
>> discover the best way to meet my requirements.  i am hoping someone will
>> have some ideas.  This could be implemented in one of the other scripting
>> language if needed.
>>
>> What i am trying to do is reprogram a remote device via the phone.  The
>> device takes commands in the form of DTMF tones and responds in different
>> DTMF tones depending on success or failure. An example of the flow would be
>>
>> freeswitch -> 342523#
>> device -> 1
>> freeswitch -> 356789#
>> device ->2
>>
>> device always responds with one digit and freeswitch sends many.
>>
>> I was hoping to do this with a database and mod_managed but i can't
>> workout how to send the DTMF in mod_managed unless i user audio files for it
>> which seems to be the wrong way to go to me. While i would prefer a
>> mod_managed solution i will take anything i can find.
>>
>>
>>
>> Chris
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From garrison at codefix.net  Thu May 13 18:11:56 2010
From: garrison at codefix.net (Garrison Hoffman)
Date: Thu, 13 May 2010 21:11:56 -0400
Subject: [Freeswitch-users] Recommendations for adapter replacement ...
In-Reply-To: <0ECBA35F-00F6-4D0F-B7A7-427DE65C6B9E@gmail.com>
References: <1273761473.1407.26.camel@strontium>
	<1273771929.1407.124.camel@strontium>
	<0ECBA35F-00F6-4D0F-B7A7-427DE65C6B9E@gmail.com>
Message-ID: <1273799516.1407.305.camel@strontium>

On Thu, 2010-05-13 at 20:42 +0200, David Ponzone wrote:

> just to be sure: does your device set rport in the INVITE ?

Never mind, I was an idiot. Trying to address more than one issue at a
time got me horribly confused; I seem to have been thinking that I could
rout the same public IP:Port to two different internal devices without
doing anything special. What amuses me now is that the configuration
worked slightly, but only for whichever device was favored by my local
router (or most recently in it's packet queue).

All is well now (except that I'm occasionally accused of having a
Sonus). I do still plan to get a new ATA, probably the Patton SmartLink
4022.

-gh



From steveu at coppice.org  Thu May 13 18:36:09 2010
From: steveu at coppice.org (Steve Underwood)
Date: Fri, 14 May 2010 09:36:09 +0800
Subject: [Freeswitch-users] Recommendations for adapter replacement ...
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E9A06B10@prato.activenetwerx.local>
References: <1273761473.1407.26.camel@strontium>	<1273771929.1407.124.camel@strontium>
	<CA5A491E9DEFBE4CB777DE97E21575E9A06B10@prato.activenetwerx.local>
Message-ID: <4BECA909.40808@coppice.org>

On 05/14/2010 01:55 AM, Joseph L. Casale wrote:
>> Naturally I'll be comparing them against Peder's
>> recommendation of the SPA2102, which has moved to the top of my list (at
>> least until I can get price info from Patton).
>>      
> Buddy, my experience is that device works well for fax, but voice
> is bad on it. It echoes like hell and support is what you expect
> from a consumer grade product...
>    
The SPA3102 has serious echo problems with its PSTN port, but its FXS 
port and the SPA2102 don't seem to give people echo problems. FAX, on 
the other hand, can be a problem with the SPA2102 and SPA3102. Their 
T.38 implementation is quirky, and often fails when the lead from the 
ATA to the FAX machine is short (say 1m to 2m). A longer lead usually 
solves this. I have no idea of the cause.

Steve




From bwibowo at gmail.com  Thu May 13 18:36:19 2010
From: bwibowo at gmail.com (Budi wibowo)
Date: Fri, 14 May 2010 01:36:19 +0000
Subject: [Freeswitch-users] mod_radius_cdr invalid ELF header
In-Reply-To: <62B274BC-8F33-48E3-9BC6-6D135ED773A8@freeswitch.org>
References: <AANLkTikcM7B-SzB7t89dyMQp6Ivhbbo3OgHTafyDYESg@mail.gmail.com><62B274BC-8F33-48E3-9BC6-6D135ED773A8@freeswitch.org>
Message-ID: <148275174-1273800981-cardhu_decombobulator_blackberry.rim.net-1349932095-@bda057.bisx.prodap.on.blackberry>

Yes correct just copy, no error right and loaded successfully. 
Still need more time to explore and prepare the radius server. 
After activate the mod_radius_cdr it override current routing rules, will have many question later :)
TIA

Budi
-----Original Message-----
From: Brian West <brian at freeswitch.org>
Date: Thu, 13 May 2010 17:01:26 
To: <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] mod_radius_cdr invalid ELF header

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




From helmut.kuper at ewetel.de  Thu May 13 23:12:28 2010
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Fri, 14 May 2010 08:12:28 +0200
Subject: [Freeswitch-users] mod_openzap, sangoma, FAX and HW-HEC
In-Reply-To: <AANLkTikUPqLyeeToYSj9SeJEdkz5cLsMT6_r6ZjEJyc3@mail.gmail.com>
References: <4BEADBBC.5090802@ewetel.de>	<AANLkTiksacBjB7uuCzZidLX5PxVFmNsE7FHNskSVjhnM@mail.gmail.com>	<AANLkTimVaDRo7ypzZu4YmgeVVMdV-sQ4TEocJOHaFv-F@mail.gmail.com>
	<AANLkTikUPqLyeeToYSj9SeJEdkz5cLsMT6_r6ZjEJyc3@mail.gmail.com>
Message-ID: <4BECE9CC.9080202@ewetel.de>

Hello Moises,

thx. Well I saw a hint for noise reduction capabilities of A104d 
somewhere in the logs during debugging. And it's easy to suppose that 
fax data could be interpreted as noise. On the other hand I'm a bit 
confused because the Fax is transmitted in a digital way, so where 
should the noise comes from (except the quantization noise).


Oh, and I use wanpipe "WANPIPE Release: 3.5.10". Is this version already 
too old?

I will try to activated this new work around to disable HEC per channel 
on demand.

Thx alot to both of you.

regards
helmut



On 12.05.2010 21:47, Moises Silva wrote:
> Sorry, the setting should be under [wanpipex] in wanpipex.conf file and
> NOT in the [wxgx] section.
>
> Moises Silva
> Senior Software Engineer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
> 9T3 Canada
> t. 1 905 474 1990 x 128 | e. moy at sangoma.com <mailto:moy at sangoma.com>


From wangdq.no1 at gmail.com  Fri May 14 00:04:27 2010
From: wangdq.no1 at gmail.com (daqiang wang)
Date: Fri, 14 May 2010 15:04:27 +0800
Subject: [Freeswitch-users] openvox PRI card for freeswitch
Message-ID: <AANLkTilvzyhuyQO09FyNv8QIwQNAgDlUpPA6YbukbsUH@mail.gmail.com>

hello:
  I test OpenVox Card :  for 4X4E1 Pri Card.
  but when the calls> 60 . the channel answer slowly ,and some errs occured.

  do you test the PRI card. ?  and why so many errors ?
      thanks .
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From mike at jerris.com  Fri May 14 00:21:31 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 14 May 2010 03:21:31 -0400
Subject: [Freeswitch-users] openvox PRI card for freeswitch
In-Reply-To: <AANLkTilvzyhuyQO09FyNv8QIwQNAgDlUpPA6YbukbsUH@mail.gmail.com>
References: <AANLkTilvzyhuyQO09FyNv8QIwQNAgDlUpPA6YbukbsUH@mail.gmail.com>
Message-ID: <321755B2-85FD-4BB6-B6D8-25BFBC0C5AA6@jerris.com>

We can't comment on your errors due to the fact that you didn't supply the errors.

Mike

On May 14, 2010, at 3:04 AM, daqiang wang wrote:

> hello:
>   I test OpenVox Card :  for 4X4E1 Pri Card. 
>   but when the calls> 60 . the channel answer slowly ,and some errs occured. 
>   do you test the PRI card. ?  and why so many errors ? 



From azatek0 at gmail.com  Fri May 14 00:29:18 2010
From: azatek0 at gmail.com (Aza Tek)
Date: Fri, 14 May 2010 09:29:18 +0200
Subject: [Freeswitch-users] ESL vs Mod_XML_Curl
Message-ID: <AANLkTilQjjBSKWuiYjXAklCGXytYkAAPMd8BWjHUL0z8@mail.gmail.com>

Hi All

I'm a bit undecided on the best approach for a web-based FreeSWITCH
application, between the above two.
Any advice would be most appreciated.

Thanks
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From david.ponzone at gmail.com  Fri May 14 01:24:29 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Fri, 14 May 2010 10:24:29 +0200
Subject: [Freeswitch-users] ESL vs Mod_XML_Curl
In-Reply-To: <AANLkTilQjjBSKWuiYjXAklCGXytYkAAPMd8BWjHUL0z8@mail.gmail.com>
References: <AANLkTilQjjBSKWuiYjXAklCGXytYkAAPMd8BWjHUL0z8@mail.gmail.com>
Message-ID: <8250325B-1AFD-42A2-9DBE-08F2D669389F@gmail.com>

Aza,

I think you got confused.

ESL and mod_xml_curl are not doing the same thing.
ESL is a way to control FreeSWITCH through the even socket (you can  
send commands like call this number, listen to DTMF, play audio file,  
etc...).

Mod_xml_curl is "just" a way to have your config, dialplan and  
directory stored in a DB, that FS will access through a HTTP/XML  
middleware.
You can't write applications with that.

The alternative to ESL is to use scripting in the dialplan (mod_lua,  
mod_python, etc...). A such application is generally put somewhere in  
the dialplan, so it is triggered because FS executes an extension.
You can also execute it from CLI (with luarun for instance).

As you say that you want to write a web-based app, I guess you should  
go for php/ESL.
Playing with that, you will perhaps find exactly what suits you.

To install php/ESL support, you need to compile FreeSWITCH with:
./configure --with-php
make install

and after that:
cd libs/esl
make phpmod-install

(php-devel is a required package)

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 14/05/2010 ? 09:29, Aza Tek a ?crit :

> Hi All
>
> I'm a bit undecided on the best approach for a web-based FreeSWITCH  
> application, between the above two.
> Any advice would be most appreciated.
>
> Thanks
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From krzysztofdrewicz at gmail.com  Fri May 14 02:24:02 2010
From: krzysztofdrewicz at gmail.com (Krzysztof Drewicz)
Date: Fri, 14 May 2010 11:24:02 +0200
Subject: [Freeswitch-users] openvox PRI card for freeswitch
In-Reply-To: <321755B2-85FD-4BB6-B6D8-25BFBC0C5AA6@jerris.com>
References: <AANLkTilvzyhuyQO09FyNv8QIwQNAgDlUpPA6YbukbsUH@mail.gmail.com>
	<321755B2-85FD-4BB6-B6D8-25BFBC0C5AA6@jerris.com>
Message-ID: <AANLkTilaF8g9io8WICUQrO7cWRmU4FPScDlIX3uyIFiC@mail.gmail.com>

2010/5/14 Michael Jerris <mike at jerris.com>:
> We can't comment on your errors due to the fact that you didn't supply the errors.
>
> Mike
>
> On May 14, 2010, at 3:04 AM, daqiang wang wrote:
>
>> hello:
>> ? I test OpenVox Card : ?for 4X4E1 Pri Card.
>> ? but when the calls> 60 . the channel answer slowly ,and some errs occured.
>> ? do you test the PRI card. ? ?and why so many errors ?

Also give us some insignt how you handle calls for example you receive
from SS7 and put into H323 with G.123 transcoding? :)
Any hw specs on your machine?

btw. i was unable to route 4 times 4 port open vox cars with pri
signalling (no sip) so no transcoding to / from RTP. Simply it was too
much for dual core Xeon with 3.6 ghz two years ago. (middle level
server from hp).

fact that you could put 4 cards into one server does not mean that you
could run every application and every appliance that you could think
of.


From krzysztofdrewicz at gmail.com  Fri May 14 02:31:36 2010
From: krzysztofdrewicz at gmail.com (Krzysztof Drewicz)
Date: Fri, 14 May 2010 11:31:36 +0200
Subject: [Freeswitch-users] mod_openzap, sangoma, FAX and HW-HEC
In-Reply-To: <4BECE9CC.9080202@ewetel.de>
References: <4BEADBBC.5090802@ewetel.de>
	<AANLkTiksacBjB7uuCzZidLX5PxVFmNsE7FHNskSVjhnM@mail.gmail.com>
	<AANLkTimVaDRo7ypzZu4YmgeVVMdV-sQ4TEocJOHaFv-F@mail.gmail.com>
	<AANLkTikUPqLyeeToYSj9SeJEdkz5cLsMT6_r6ZjEJyc3@mail.gmail.com>
	<4BECE9CC.9080202@ewetel.de>
Message-ID: <AANLkTimPS8ygPYTMKqozBndkRNLGigj_loAryLtKQXuC@mail.gmail.com>

2010/5/14 Helmut Kuper <helmut.kuper at ewetel.de>:
> Hello Moises,
>
> thx. Well I saw a hint for noise reduction capabilities of A104d
> somewhere in the logs during debugging. And it's easy to suppose that
> fax data could be interpreted as noise. On the other hand I'm a bit
> confused because the Fax is transmitted in a digital way, so where
> should the noise comes from (except the quantization noise).

Noise could come from analog line "on the other side", for example
this is 2 phones (one being facsimile machine) put in parallel to one
pair of copper line.
I run some big faxserver farm (like having 20+ channels up most of the
9-17 on one machine and having two machines in two different
locations)  and prefer to use some openvox cards without any EC
(hardware or software). with A102d i had some problems that caused
need to reboot server with a reset button every two-three weeks.

btw: i use dedicated machine just for faxing, for you disable 'on
demant' might work better.

what software you use for fax? (iaxmodem/ t38modem + hylafax or you do
faxes in FS ?) and what speed you manage to get (14 400 bps or some
more)?


From wasim at convergence.pk  Fri May 14 03:06:21 2010
From: wasim at convergence.pk (Wasim Baig)
Date: Fri, 14 May 2010 15:06:21 +0500
Subject: [Freeswitch-users] openvox PRI card for freeswitch
In-Reply-To: <AANLkTilaF8g9io8WICUQrO7cWRmU4FPScDlIX3uyIFiC@mail.gmail.com>
References: <AANLkTilvzyhuyQO09FyNv8QIwQNAgDlUpPA6YbukbsUH@mail.gmail.com> 
	<321755B2-85FD-4BB6-B6D8-25BFBC0C5AA6@jerris.com>
	<AANLkTilaF8g9io8WICUQrO7cWRmU4FPScDlIX3uyIFiC@mail.gmail.com>
Message-ID: <AANLkTimpNQ9pS-HPacFtg8RRpTbhIqpXsGWCYQ532Uor@mail.gmail.com>

also try removing all but 1 of the cards, and see if you still have the
issue with 60+ calls

if not, then add 1 card, and retest ..

-wasim

On Fri, May 14, 2010 at 14:24, Krzysztof Drewicz <krzysztofdrewicz at gmail.com
> wrote:

> 2010/5/14 Michael Jerris <mike at jerris.com>:
> > We can't comment on your errors due to the fact that you didn't supply
> the errors.
> >
> > Mike
> >
> > On May 14, 2010, at 3:04 AM, daqiang wang wrote:
> >
> >> hello:
> >>   I test OpenVox Card :  for 4X4E1 Pri Card.
> >>   but when the calls> 60 . the channel answer slowly ,and some errs
> occured.
> >>   do you test the PRI card. ?  and why so many errors ?
>
> Also give us some insignt how you handle calls for example you receive
> from SS7 and put into H323 with G.123 transcoding? :)
> Any hw specs on your machine?
>
> btw. i was unable to route 4 times 4 port open vox cars with pri
> signalling (no sip) so no transcoding to / from RTP. Simply it was too
> much for dual core Xeon with 3.6 ghz two years ago. (middle level
> server from hp).
>
> fact that you could put 4 cards into one server does not mean that you
> could run every application and every appliance that you could think
> of.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
wasim h. baig | principal consultant | convergence pk | +92 30 0850 8070 |
peace be upon you ...
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From david.ponzone at gmail.com  Fri May 14 03:18:52 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Fri, 14 May 2010 12:18:52 +0200
Subject: [Freeswitch-users] openvox PRI card for freeswitch
In-Reply-To: <AANLkTilaF8g9io8WICUQrO7cWRmU4FPScDlIX3uyIFiC@mail.gmail.com>
References: <AANLkTilvzyhuyQO09FyNv8QIwQNAgDlUpPA6YbukbsUH@mail.gmail.com>
	<321755B2-85FD-4BB6-B6D8-25BFBC0C5AA6@jerris.com>
	<AANLkTilaF8g9io8WICUQrO7cWRmU4FPScDlIX3uyIFiC@mail.gmail.com>
Message-ID: <69A6B994-FCE3-497D-A43E-2B4F83419754@gmail.com>

Krzysztof,

May I use the opportunity to ask your opinion on those cards ?
They are strangely inexpensive (particularly the D110/115), and I  
can't find any mention of HW EC on their website.
Can you confirm this ?
Any issue using it with FreeSWITCH ?

Thank you

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 14/05/2010 ? 11:24, Krzysztof Drewicz a ?crit :

> 2010/5/14 Michael Jerris <mike at jerris.com>:
>> We can't comment on your errors due to the fact that you didn't  
>> supply the errors.
>>
>> Mike
>>
>> On May 14, 2010, at 3:04 AM, daqiang wang wrote:
>>
>>> hello:
>>>   I test OpenVox Card :  for 4X4E1 Pri Card.
>>>   but when the calls> 60 . the channel answer slowly ,and some  
>>> errs occured.
>>>   do you test the PRI card. ?  and why so many errors ?
>
> Also give us some insignt how you handle calls for example you receive
> from SS7 and put into H323 with G.123 transcoding? :)
> Any hw specs on your machine?
>
> btw. i was unable to route 4 times 4 port open vox cars with pri
> signalling (no sip) so no transcoding to / from RTP. Simply it was too
> much for dual core Xeon with 3.6 ghz two years ago. (middle level
> server from hp).
>
> fact that you could put 4 cards into one server does not mean that you
> could run every application and every appliance that you could think
> of.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Fri May 14 03:23:11 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 14 May 2010 06:23:11 -0400
Subject: [Freeswitch-users] openvox PRI card for freeswitch
In-Reply-To: <AANLkTilaF8g9io8WICUQrO7cWRmU4FPScDlIX3uyIFiC@mail.gmail.com>
References: <AANLkTilvzyhuyQO09FyNv8QIwQNAgDlUpPA6YbukbsUH@mail.gmail.com>
	<321755B2-85FD-4BB6-B6D8-25BFBC0C5AA6@jerris.com>
	<AANLkTilaF8g9io8WICUQrO7cWRmU4FPScDlIX3uyIFiC@mail.gmail.com>
Message-ID: <7C84BEAD-691E-4403-902A-E977B0EBDD00@jerris.com>

How did this thread go from PRI to ss7 and h323 ?  Is there a connection between these?

On May 14, 2010, at 5:24 AM, Krzysztof Drewicz wrote:

> 2010/5/14 Michael Jerris <mike at jerris.com>:
>> We can't comment on your errors due to the fact that you didn't supply the errors.
>> 
>> Mike
>> 
>> On May 14, 2010, at 3:04 AM, daqiang wang wrote:
>> 
>>> hello:
>>>   I test OpenVox Card :  for 4X4E1 Pri Card.
>>>   but when the calls> 60 . the channel answer slowly ,and some errs occured.
>>>   do you test the PRI card. ?  and why so many errors ?
> 
> Also give us some insignt how you handle calls for example you receive
> from SS7 and put into H323 with G.123 transcoding? :)
> Any hw specs on your machine?
> 
> btw. i was unable to route 4 times 4 port open vox cars with pri
> signalling (no sip) so no transcoding to / from RTP. Simply it was too
> much for dual core Xeon with 3.6 ghz two years ago. (middle level
> server from hp).
> 
> fact that you could put 4 cards into one server does not mean that you
> could run every application and every appliance that you could think
> of.
> 



From krzysztofdrewicz at gmail.com  Fri May 14 05:05:01 2010
From: krzysztofdrewicz at gmail.com (Krzysztof Drewicz)
Date: Fri, 14 May 2010 14:05:01 +0200
Subject: [Freeswitch-users] openvox PRI card for freeswitch
In-Reply-To: <69A6B994-FCE3-497D-A43E-2B4F83419754@gmail.com>
References: <AANLkTilvzyhuyQO09FyNv8QIwQNAgDlUpPA6YbukbsUH@mail.gmail.com>
	<321755B2-85FD-4BB6-B6D8-25BFBC0C5AA6@jerris.com>
	<AANLkTilaF8g9io8WICUQrO7cWRmU4FPScDlIX3uyIFiC@mail.gmail.com>
	<69A6B994-FCE3-497D-A43E-2B4F83419754@gmail.com>
Message-ID: <AANLkTikxdU67WA3MWO1TZR_8FHxZIxZth9h0u-hg_9Ap@mail.gmail.com>

2010/5/14 David Ponzone <david.ponzone at gmail.com>:
> Krzysztof,
> May I use the opportunity to ask your opinion on those cards ?
> They are strangely inexpensive (particularly the D110/115), and I can't find
> any mention of HW EC on their website.

The one-port are not very good (no hw ec, even as an option). But for
simple use as faxserver without any 'big' traffic they seem to work
fine.
But still, one card per server :(

> Can you confirm this ?
> Any issue using it with FreeSWITCH ?

Hm, whould recomend the 4 port, they got hw ec (as an option but buy
hw ec with card, even if you plan to use it afterwards, some times
there is card with older firmware and hw ec has new firmware and they
don't match).
Card are recognized as Digium cards, they work :) just don't put too
much traffic on them (one 4 port card per one good box seems to be
OK).

I buy them from some local internet-based shop and never had any
problem (like card not working or not compatible with server mainboard
etc...).


From azatek0 at gmail.com  Fri May 14 05:22:46 2010
From: azatek0 at gmail.com (Aza Tek)
Date: Fri, 14 May 2010 14:22:46 +0200
Subject: [Freeswitch-users] ESL vs Mod_XML_Curl
In-Reply-To: <8250325B-1AFD-42A2-9DBE-08F2D669389F@gmail.com>
References: <AANLkTilQjjBSKWuiYjXAklCGXytYkAAPMd8BWjHUL0z8@mail.gmail.com> 
	<8250325B-1AFD-42A2-9DBE-08F2D669389F@gmail.com>
Message-ID: <AANLkTinx-6m04dOMaHMW92Tz76_JyABuuJ7-scWbpSLK@mail.gmail.com>

Thanks David that was quite informative. Is there a particular benefit to
having something like a Dialplan stored in a database?


On Fri, May 14, 2010 at 10:24 AM, David Ponzone <david.ponzone at gmail.com>wrote:

> Aza,
>
> I think you got confused.
>
> ESL and mod_xml_curl are not doing the same thing.
> ESL is a way to control FreeSWITCH through the even socket (you can send
> commands like call this number, listen to DTMF, play audio file, etc...).
>
> Mod_xml_curl is "just" a way to have your config, dialplan and directory
> stored in a DB, that FS will access through a HTTP/XML middleware.
> You can't write applications with that.
>
> The alternative to ESL is to use scripting in the dialplan (mod_lua,
> mod_python, etc...). A such application is generally put somewhere in the
> dialplan, so it is triggered because FS executes an extension.
> You can also execute it from CLI (with luarun for instance).
>
> As you say that you want to write a web-based app, I guess you should go
> for php/ESL.
> Playing with that, you will perhaps find exactly what suits you.
>
> To install php/ESL support, you need to compile FreeSWITCH with:
> ./configure --with-php
> make install
>
> and after that:
> cd libs/esl
> make phpmod-install
>
> (php-devel is a required package)
>
> David Ponzone  Direction Technique
>  email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 14/05/2010 ? 09:29, Aza Tek a ?crit :
>
> Hi All
>
> I'm a bit undecided on the best approach for a web-based FreeSWITCH
> application, between the above two.
> Any advice would be most appreciated.
>
> Thanks
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at gmail.com  Fri May 14 05:55:54 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Fri, 14 May 2010 14:55:54 +0200
Subject: [Freeswitch-users] ESL vs Mod_XML_Curl
In-Reply-To: <AANLkTinx-6m04dOMaHMW92Tz76_JyABuuJ7-scWbpSLK@mail.gmail.com>
References: <AANLkTilQjjBSKWuiYjXAklCGXytYkAAPMd8BWjHUL0z8@mail.gmail.com>
	<8250325B-1AFD-42A2-9DBE-08F2D669389F@gmail.com>
	<AANLkTinx-6m04dOMaHMW92Tz76_JyABuuJ7-scWbpSLK@mail.gmail.com>
Message-ID: <1277E87F-B8D7-43C8-964F-8B83E33EBDD9@gmail.com>

Well, it's dynamical, and it can be shared among multiple FS instances.

When you use FS only as a switch between trunks, it's not really  
interesting, but if you plan to have lots of users registered on FS,  
it may help.
Only you can decide if you need a static or dynamic dialplan.
And if you need to register lots of users, it's pretty sure you'll  
want to store them in a DB...

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 14/05/2010 ? 14:22, Aza Tek a ?crit :

> Thanks David that was quite informative. Is there a particular  
> benefit to having something like a Dialplan stored in a database?
>
>
> On Fri, May 14, 2010 at 10:24 AM, David Ponzone <david.ponzone at gmail.com 
> > wrote:
> Aza,
>
> I think you got confused.
>
> ESL and mod_xml_curl are not doing the same thing.
> ESL is a way to control FreeSWITCH through the even socket (you can  
> send commands like call this number, listen to DTMF, play audio  
> file, etc...).
>
> Mod_xml_curl is "just" a way to have your config, dialplan and  
> directory stored in a DB, that FS will access through a HTTP/XML  
> middleware.
> You can't write applications with that.
>
> The alternative to ESL is to use scripting in the dialplan (mod_lua,  
> mod_python, etc...). A such application is generally put somewhere  
> in the dialplan, so it is triggered because FS executes an extension.
> You can also execute it from CLI (with luarun for instance).
>
> As you say that you want to write a web-based app, I guess you  
> should go for php/ESL.
> Playing with that, you will perhaps find exactly what suits you.
>
> To install php/ESL support, you need to compile FreeSWITCH with:
> ./configure --with-php
> make install
>
> and after that:
> cd libs/esl
> make phpmod-install
>
> (php-devel is a required package)
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et  
> ?tablis ? l'intention exclusive de ses destinataires. Toute  
> utilisation ou diffusion non autoris?e est interdite. Tout message  
> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou  
> falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le  
> d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
>
> Le 14/05/2010 ? 09:29, Aza Tek a ?crit :
>
>> Hi All
>>
>> I'm a bit undecided on the best approach for a web-based FreeSWITCH  
>> application, between the above two.
>> Any advice would be most appreciated.
>>
>> Thanks
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From david.ponzone at gmail.com  Fri May 14 05:59:23 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Fri, 14 May 2010 14:59:23 +0200
Subject: [Freeswitch-users] openvox PRI card for freeswitch
In-Reply-To: <AANLkTikxdU67WA3MWO1TZR_8FHxZIxZth9h0u-hg_9Ap@mail.gmail.com>
References: <AANLkTilvzyhuyQO09FyNv8QIwQNAgDlUpPA6YbukbsUH@mail.gmail.com>
	<321755B2-85FD-4BB6-B6D8-25BFBC0C5AA6@jerris.com>
	<AANLkTilaF8g9io8WICUQrO7cWRmU4FPScDlIX3uyIFiC@mail.gmail.com>
	<69A6B994-FCE3-497D-A43E-2B4F83419754@gmail.com>
	<AANLkTikxdU67WA3MWO1TZR_8FHxZIxZth9h0u-hg_9Ap@mail.gmail.com>
Message-ID: <E867C025-C339-49EA-8D69-25EB3DB58233@gmail.com>

4 ports is not an option as my goal is to find a cheaper alternative  
to the Sangoma A101D/DE so I can build a reliable but cheap 1PRI/SIP  
gateway.

It seems in the end, Sangoma has no alternative.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 14/05/2010 ? 14:05, Krzysztof Drewicz a ?crit :

> 2010/5/14 David Ponzone <david.ponzone at gmail.com>:
>> Krzysztof,
>> May I use the opportunity to ask your opinion on those cards ?
>> They are strangely inexpensive (particularly the D110/115), and I  
>> can't find
>> any mention of HW EC on their website.
>
> The one-port are not very good (no hw ec, even as an option). But for
> simple use as faxserver without any 'big' traffic they seem to work
> fine.
> But still, one card per server :(
>
>> Can you confirm this ?
>> Any issue using it with FreeSWITCH ?
>
> Hm, whould recomend the 4 port, they got hw ec (as an option but buy
> hw ec with card, even if you plan to use it afterwards, some times
> there is card with older firmware and hw ec has new firmware and they
> don't match).
> Card are recognized as Digium cards, they work :) just don't put too
> much traffic on them (one 4 port card per one good box seems to be
> OK).
>
> I buy them from some local internet-based shop and never had any
> problem (like card not working or not compatible with server mainboard
> etc...).
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From krzysztofdrewicz at gmail.com  Fri May 14 06:08:13 2010
From: krzysztofdrewicz at gmail.com (Krzysztof Drewicz)
Date: Fri, 14 May 2010 15:08:13 +0200
Subject: [Freeswitch-users] openvox PRI card for freeswitch
In-Reply-To: <E867C025-C339-49EA-8D69-25EB3DB58233@gmail.com>
References: <AANLkTilvzyhuyQO09FyNv8QIwQNAgDlUpPA6YbukbsUH@mail.gmail.com>
	<321755B2-85FD-4BB6-B6D8-25BFBC0C5AA6@jerris.com>
	<AANLkTilaF8g9io8WICUQrO7cWRmU4FPScDlIX3uyIFiC@mail.gmail.com>
	<69A6B994-FCE3-497D-A43E-2B4F83419754@gmail.com>
	<AANLkTikxdU67WA3MWO1TZR_8FHxZIxZth9h0u-hg_9Ap@mail.gmail.com>
	<E867C025-C339-49EA-8D69-25EB3DB58233@gmail.com>
Message-ID: <AANLkTinuXnpZv9ogTJ79R4Z9Rwl5ekBhullQyRZ_nG07@mail.gmail.com>

2010/5/14 David Ponzone <david.ponzone at gmail.com>:
> 4 ports is not an option as my goal is to find a cheaper alternative to the
> Sangoma A101D/DE so I can build a reliable but cheap 1PRI/SIP gateway.
> It seems in the end, Sangoma has no alternative.


Use open vox with 2 port, it's cheap.


From tculjaga at gmail.com  Fri May 14 06:15:53 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 14 May 2010 15:15:53 +0200
Subject: [Freeswitch-users] fetch caller profile variables
In-Reply-To: <AANLkTim4KO8XRwl-n9-tq-BSFpwC2Jds1H752Rh9uNIk@mail.gmail.com>
References: <AANLkTinXlIUddb_xAWh1u3DLpAuuWQhisOOA8ygWLGnU@mail.gmail.com>
	<AANLkTim4KO8XRwl-n9-tq-BSFpwC2Jds1H752Rh9uNIk@mail.gmail.com>
Message-ID: <AANLkTiln1I_oiqgsi4UEKusm9ICTcuvHfsrW_2np4qyk@mail.gmail.com>

Indeed, this is how i do it ... mod_xml_cdr ... but i need to pupulate some
session variables before the CDR is generated (before on_my_reporting
event).

something like this:

      <action inline="true" application="set"
data="GW_COLLECTED_CDN=gw-collected-cdn=${DIALED_NUMBER}"/>
      <action inline="true" application="set"
data="GK_XLATED_CDN=gk-xlated-cdn=006800${DIALED_NUMBER}"/>
      <action inline="true" application="set"
data="GW_FINAL_XLATED_CDN=gw-final-xlated-cdn=ton:0,npi:0,#:=006800${DIALED_NUMBER}"/>
      <action inline="true" application="set"
data="GW_FINAL_XLATED_CGN=gw-final-xlated-cgn=ton:2,npi:1,pi:0,si:1,#:=${DIALED_NUMBER}"/>



i was thinking about something like this but not working correctly:


      <action application="export"
data="nolocal:api_hangup_hook=uuid_setvar_multi ${uuid}
CONNECT_TIME=h323-connect-time=${answered
_time}; DISCONNECT_TIME=h323-disconnect-time=${hangup_time};
RELEASE_CAUSE=h323-disconnect-cause=${hangup_cause_q850}; ALERT_TIME=al
ert-timepoint=${created_time}; "/>



freeswitch at cxss01> 2010-05-14 15:56:49.697513 [DEBUG] mod_xml_cdr.c:533
RADIUS Accounting OK
2010-05-14 15:56:49.697513 [NOTICE] switch_core_session.c:1945 Execute
export(nolocal:api_hangup_hook=uuid_setvar_multi ${uuid}
CONNECT_TIME=h323-connect-time=${answered_time};
DISCONNECT_TIME=h323-disconnect-time=${hangup_time};
RELEASE_CAUSE=h323-disconnect-cause=${hangup_cause_q850};
ALERT_TIME=alert-timepoint=${created_time}; )
EXECUTE sofia/external/0038515492122 at 195.88.212.41export(nolocal:api_hangup_hook=uuid_setvar_multi
ff27c7c6-d37b-4ee5-8fb5-0e3cb36ad603
CONNECT_TIME=h323-connect-time=1273845357249457;
DISCONNECT_TIME=h323-disconnect-time=; RELEASE_CAUSE=h323-disconnect-cause=;
ALERT_TIME=alert-timepoint=1273845357240520; )
2010-05-14 15:56:49.697513 [DEBUG] mod_dptools.c:898 EXPORT (REMOTE ONLY)
[api_hangup_hook]=[uuid_setvar_multi ff27c7c6-d37b-4ee5-8fb5-0e3cb36ad603
CONNECT_TIME=h323-connect-time=1273845357249457;
DISCONNECT_TIME=h323-disconnect-time=; RELEASE_CAUSE=h323-disconnect-cause=;
ALERT_TIME=alert-timepoint=1273845357240520; ]





when can i set this vars? What event can i use ?



also, i need to trigger some application after bridge but before ringing ...
any suggestion ?


Tihomir.




On Thu, May 13, 2010 at 4:39 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> answer_stamp is a regular channel variable not a caller profile variable
> ${answer_stamp} is correct but its only set once the channel is hungup
>
> billing is best done in a separate process from the dialplan on the backend
> via EVENTS or CDR
>
>
> On Thu, May 13, 2010 at 4:36 AM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>> hello,
>>
>> i need to fetch a caller profile variable (e.g. answer_stamp) from a
>> dialplan, can anyone help?
>>
>>
>> i tried with:
>>
>> <action application="set" data="TEST=answer_stamp"/>
>> <action inline="true" application="set" data="TEST=answer_stamp"/>
>> <action inline="true" application="set" data="TEST=${answer_stamp}"/>
>>
>>
>> but it doesn't help...
>>
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveu at coppice.org  Fri May 14 06:22:29 2010
From: steveu at coppice.org (Steve Underwood)
Date: Fri, 14 May 2010 21:22:29 +0800
Subject: [Freeswitch-users] mod_openzap, sangoma, FAX and HW-HEC
In-Reply-To: <AANLkTikUPqLyeeToYSj9SeJEdkz5cLsMT6_r6ZjEJyc3@mail.gmail.com>
References: <4BEADBBC.5090802@ewetel.de>	<AANLkTiksacBjB7uuCzZidLX5PxVFmNsE7FHNskSVjhnM@mail.gmail.com>	<AANLkTimVaDRo7ypzZu4YmgeVVMdV-sQ4TEocJOHaFv-F@mail.gmail.com>
	<AANLkTikUPqLyeeToYSj9SeJEdkz5cLsMT6_r6ZjEJyc3@mail.gmail.com>
Message-ID: <4BED4E95.8090508@coppice.org>

Hi Moises,

Do you know exactly what the noise reduction does? In an echo canceller 
context I can see a couple of possibilities. I think all of them are 
pretty horrible for what they might do to modem signals. Maybe this has 
something to do with why so many people have problems with both Sangoma 
and Digium boards that use the Octasic echo cancellers.

Steve


On 05/13/2010 03:47 AM, Moises Silva wrote:
> Sorry, the setting should be under [wanpipex] in wanpipex.conf file 
> and NOT in the [wxgx] section.
>
> Moises Silva
> Senior Software Engineer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON 
> L3R 9T3 Canada
> t. 1 905 474 1990 x 128 | e. moy at sangoma.com <mailto:moy at sangoma.com>
>
>
> On Wed, May 12, 2010 at 3:40 PM, Moises Silva <moises.silva at gmail.com 
> <mailto:moises.silva at gmail.com>> wrote:
>
>     It is implemented and should work as a work-around.
>
>     Having said that, I suggest you to contact Sangoma support
>     (techdesk at sangoma.com <mailto:techdesk at sangoma.com>), the ec chip
>     should detect fax tone and disable its echo-canceling functions.
>     However, some old versions of the driver have a noise-reduction
>     feature enabled by default which also happen to disrupt faxes.
>
>     Try adding this just below TDMV_HWEC=YES option:
>
>     HWEC_NOISE_REDUCTION_DISABLE=YES
>
>     Or upgrade to the latest driver which does this automatically.
>



From david.ponzone at gmail.com  Fri May 14 06:26:31 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Fri, 14 May 2010 15:26:31 +0200
Subject: [Freeswitch-users] openvox PRI card for freeswitch
In-Reply-To: <AANLkTinuXnpZv9ogTJ79R4Z9Rwl5ekBhullQyRZ_nG07@mail.gmail.com>
References: <AANLkTilvzyhuyQO09FyNv8QIwQNAgDlUpPA6YbukbsUH@mail.gmail.com>
	<321755B2-85FD-4BB6-B6D8-25BFBC0C5AA6@jerris.com>
	<AANLkTilaF8g9io8WICUQrO7cWRmU4FPScDlIX3uyIFiC@mail.gmail.com>
	<69A6B994-FCE3-497D-A43E-2B4F83419754@gmail.com>
	<AANLkTikxdU67WA3MWO1TZR_8FHxZIxZth9h0u-hg_9Ap@mail.gmail.com>
	<E867C025-C339-49EA-8D69-25EB3DB58233@gmail.com>
	<AANLkTinuXnpZv9ogTJ79R4Z9Rwl5ekBhullQyRZ_nG07@mail.gmail.com>
Message-ID: <DB9F5F13-72C3-48C3-BF9B-298CA4FFFF26@gmail.com>

Well, the DE210E is 999$ which is significantly more expensive than  
the Sandoma A101DE, except if I can expect a 40% discount on OpenVox  
products.
But thanks for the idea anyway, I'll check this out.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 14/05/2010 ? 15:08, Krzysztof Drewicz a ?crit :

> 2010/5/14 David Ponzone <david.ponzone at gmail.com>:
>> 4 ports is not an option as my goal is to find a cheaper  
>> alternative to the
>> Sangoma A101D/DE so I can build a reliable but cheap 1PRI/SIP  
>> gateway.
>> It seems in the end, Sangoma has no alternative.
>
>
> Use open vox with 2 port, it's cheap.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From helmut.kuper at ewetel.de  Fri May 14 06:53:20 2010
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Fri, 14 May 2010 15:53:20 +0200
Subject: [Freeswitch-users] mod_openzap, sangoma, FAX and HW-HEC
In-Reply-To: <AANLkTimPS8ygPYTMKqozBndkRNLGigj_loAryLtKQXuC@mail.gmail.com>
References: <4BEADBBC.5090802@ewetel.de>	<AANLkTiksacBjB7uuCzZidLX5PxVFmNsE7FHNskSVjhnM@mail.gmail.com>	<AANLkTimVaDRo7ypzZu4YmgeVVMdV-sQ4TEocJOHaFv-F@mail.gmail.com>	<AANLkTikUPqLyeeToYSj9SeJEdkz5cLsMT6_r6ZjEJyc3@mail.gmail.com>	<4BECE9CC.9080202@ewetel.de>
	<AANLkTimPS8ygPYTMKqozBndkRNLGigj_loAryLtKQXuC@mail.gmail.com>
Message-ID: <4BED55D0.4080603@ewetel.de>

Hello, erm, Krzysztof (?) (broke my eyes first, then my tongue and at 
last my fingers :D :D ;-) )

well I use FS simply as a (G711a) passthrough gateway to good old analog 
fax devices connected via ATAs to FS. Bitrates are < 14400. Quite 
simple, works currently, but have to waive the HEC, until I tried the 
workaround. I will have in the end up to 60 fax devices, maybe a 
dedicated machine is needed then. But currently I have only two devices.


regards
helmut


On 14.05.2010 11:31, Krzysztof Drewicz wrote:
> 2010/5/14 Helmut Kuper<helmut.kuper at ewetel.de>:
>> Hello Moises,
>>
>> thx. Well I saw a hint for noise reduction capabilities of A104d
>> somewhere in the logs during debugging. And it's easy to suppose that
>> fax data could be interpreted as noise. On the other hand I'm a bit
>> confused because the Fax is transmitted in a digital way, so where
>> should the noise comes from (except the quantization noise).
>
> Noise could come from analog line "on the other side", for example
> this is 2 phones (one being facsimile machine) put in parallel to one
> pair of copper line.
> I run some big faxserver farm (like having 20+ channels up most of the
> 9-17 on one machine and having two machines in two different
> locations)  and prefer to use some openvox cards without any EC
> (hardware or software). with A102d i had some problems that caused
> need to reboot server with a reset button every two-three weeks.
>
> btw: i use dedicated machine just for faxing, for you disable 'on
> demant' might work better.
>
> what software you use for fax? (iaxmodem/ t38modem + hylafax or you do
> faxes in FS ?) and what speed you manage to get (14 400 bps or some
> more)?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From tculjaga at gmail.com  Fri May 14 07:43:49 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Fri, 14 May 2010 16:43:49 +0200
Subject: [Freeswitch-users] fetch caller profile variables
In-Reply-To: <AANLkTiln1I_oiqgsi4UEKusm9ICTcuvHfsrW_2np4qyk@mail.gmail.com>
References: <AANLkTinXlIUddb_xAWh1u3DLpAuuWQhisOOA8ygWLGnU@mail.gmail.com>
	<AANLkTim4KO8XRwl-n9-tq-BSFpwC2Jds1H752Rh9uNIk@mail.gmail.com>
	<AANLkTiln1I_oiqgsi4UEKusm9ICTcuvHfsrW_2np4qyk@mail.gmail.com>
Message-ID: <AANLkTinw48Gbf9GiwC-0icNDsJ0YPvADTklBVLh9XpFa@mail.gmail.com>

btw:


if you are wondering, yes we added radius accounting & ODBC support for
mod_xml_cdr ... both works perfectly, but radius needs some Attribute
manipulation and this is what im trying to do via dialplan :))


T.


On Fri, May 14, 2010 at 3:15 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> Indeed, this is how i do it ... mod_xml_cdr ... but i need to pupulate some
> session variables before the CDR is generated (before on_my_reporting
> event).
>
> something like this:
>
>       <action inline="true" application="set"
> data="GW_COLLECTED_CDN=gw-collected-cdn=${DIALED_NUMBER}"/>
>       <action inline="true" application="set"
> data="GK_XLATED_CDN=gk-xlated-cdn=006800${DIALED_NUMBER}"/>
>       <action inline="true" application="set"
> data="GW_FINAL_XLATED_CDN=gw-final-xlated-cdn=ton:0,npi:0,#:=006800${DIALED_NUMBER}"/>
>       <action inline="true" application="set"
> data="GW_FINAL_XLATED_CGN=gw-final-xlated-cgn=ton:2,npi:1,pi:0,si:1,#:=${DIALED_NUMBER}"/>
>
>
>
> i was thinking about something like this but not working correctly:
>
>
>       <action application="export"
> data="nolocal:api_hangup_hook=uuid_setvar_multi ${uuid}
> CONNECT_TIME=h323-connect-time=${answered
> _time}; DISCONNECT_TIME=h323-disconnect-time=${hangup_time};
> RELEASE_CAUSE=h323-disconnect-cause=${hangup_cause_q850}; ALERT_TIME=al
> ert-timepoint=${created_time}; "/>
>
>
>
> freeswitch at cxss01> 2010-05-14 15:56:49.697513 [DEBUG] mod_xml_cdr.c:533
> RADIUS Accounting OK
> 2010-05-14 15:56:49.697513 [NOTICE] switch_core_session.c:1945 Execute
> export(nolocal:api_hangup_hook=uuid_setvar_multi ${uuid}
> CONNECT_TIME=h323-connect-time=${answered_time};
> DISCONNECT_TIME=h323-disconnect-time=${hangup_time};
> RELEASE_CAUSE=h323-disconnect-cause=${hangup_cause_q850};
> ALERT_TIME=alert-timepoint=${created_time}; )
> EXECUTE sofia/external/0038515492122 at 195.88.212.41export(nolocal:api_hangup_hook=uuid_setvar_multi
> ff27c7c6-d37b-4ee5-8fb5-0e3cb36ad603
> CONNECT_TIME=h323-connect-time=1273845357249457;
> DISCONNECT_TIME=h323-disconnect-time=; RELEASE_CAUSE=h323-disconnect-cause=;
> ALERT_TIME=alert-timepoint=1273845357240520; )
> 2010-05-14 15:56:49.697513 [DEBUG] mod_dptools.c:898 EXPORT (REMOTE ONLY)
> [api_hangup_hook]=[uuid_setvar_multi ff27c7c6-d37b-4ee5-8fb5-0e3cb36ad603
> CONNECT_TIME=h323-connect-time=1273845357249457;
> DISCONNECT_TIME=h323-disconnect-time=; RELEASE_CAUSE=h323-disconnect-cause=;
> ALERT_TIME=alert-timepoint=1273845357240520; ]
>
>
>
>
>
> when can i set this vars? What event can i use ?
>
>
>
> also, i need to trigger some application after bridge but before ringing
> ... any suggestion ?
>
>
> Tihomir.
>
>
>
>
>
> On Thu, May 13, 2010 at 4:39 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> answer_stamp is a regular channel variable not a caller profile variable
>> ${answer_stamp} is correct but its only set once the channel is hungup
>>
>> billing is best done in a separate process from the dialplan on the
>> backend via EVENTS or CDR
>>
>>
>> On Thu, May 13, 2010 at 4:36 AM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>
>>> hello,
>>>
>>> i need to fetch a caller profile variable (e.g. answer_stamp) from a
>>> dialplan, can anyone help?
>>>
>>>
>>> i tried with:
>>>
>>> <action application="set" data="TEST=answer_stamp"/>
>>> <action inline="true" application="set" data="TEST=answer_stamp"/>
>>> <action inline="true" application="set" data="TEST=${answer_stamp}"/>
>>>
>>>
>>> but it doesn't help...
>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <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-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From talk2ram at gmail.com  Fri May 14 07:46:58 2010
From: talk2ram at gmail.com (ram)
Date: Fri, 14 May 2010 20:16:58 +0530
Subject: [Freeswitch-users] freeswitch RPM available for centos 5.4
Message-ID: <AANLkTinzMmrg3BqZHmpA7TeBPR_0jEYRchGuXr9dNEDZ@mail.gmail.com>

hi all

we have recently build freeswitch RPM  ( based on 13th may 2010 git) for
fusionpbx ISO

you can try installing ISO or RPMS

http://wiki.fusionpbx.com/index.php/CentOS_ISO


Ram
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From brian at freeswitch.org  Fri May 14 07:54:00 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 14 May 2010 09:54:00 -0500
Subject: [Freeswitch-users] freeswitch RPM available for centos 5.4
In-Reply-To: <AANLkTinzMmrg3BqZHmpA7TeBPR_0jEYRchGuXr9dNEDZ@mail.gmail.com>
References: <AANLkTinzMmrg3BqZHmpA7TeBPR_0jEYRchGuXr9dNEDZ@mail.gmail.com>
Message-ID: <9856D01D-A4FD-4C6F-8E01-48426E7C0557@freeswitch.org>

While I welcome your RPM's you should at the very least compile 64bit version too... we are currently working to provide, 32,64 for CentOS, Debian and Ubuntu along with nightly builds of OS X and Windows 32bit ... If anyone wishes to help configure a host in our build farm we are going to deploy later this month please email me.

Thanks,
Brian

On May 14, 2010, at 9:46 AM, ram wrote:

> hi all
>  
> we have recently build freeswitch RPM  ( based on 13th may 2010 git) for fusionpbx ISO
>  
> you can try installing ISO or RPMS
>  
> http://wiki.fusionpbx.com/index.php/CentOS_ISO
>  
>  
> Ram
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From toqeer83 at gmail.com  Fri May 14 08:09:20 2010
From: toqeer83 at gmail.com (toqeer ali)
Date: Fri, 14 May 2010 08:09:20 -0700
Subject: [Freeswitch-users] Hunting
Message-ID: <AANLkTinW51XkJQAqDdI-D8760rrVOnMfaFomGjzLH7-L@mail.gmail.com>

Hi all,

I am new to freeswitch and want to implement Hunting.

For example if call come to my freeswitch box where DID is configured to an
extension for example 1000 so if 1000 did not pick the phone then it should
be transfered to other extension or landline.

Please give me a clue... Is there is any special module for that?

Any help will highly appreciated.

Thanks

-- 
Toqeer Ali Syed

Red Hat Certified Engineer
mob:     +92 321 9059916
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From fs-list at communicatefreely.net  Fri May 14 08:49:38 2010
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Fri, 14 May 2010 11:49:38 -0400
Subject: [Freeswitch-users] Hunting
In-Reply-To: <AANLkTinW51XkJQAqDdI-D8760rrVOnMfaFomGjzLH7-L@mail.gmail.com>
References: <AANLkTinW51XkJQAqDdI-D8760rrVOnMfaFomGjzLH7-L@mail.gmail.com>
Message-ID: <4BED7112.1040701@communicatefreely.net>

Nope, no special module required.

Before you call the bridge application, set continue_on_fail=true
This will tell freeswitch to keep going in the dialplan if the first destination doesn't answer.
After the first bridge application, add other entries that take try the next destination.  You could
use the transfer app to try another extension as you described.

-Tim

toqeer ali wrote:
> Hi all,
> 
> I am new to freeswitch and want to implement Hunting.
> 
> For example if call come to my freeswitch box where DID is configured to
> an extension for example 1000 so if 1000 did not pick the phone then it
> should be transfered to other extension or landline.
> 
> Please give me a clue... Is there is any special module for that?
> 
> Any help will highly appreciated.
> 
> Thanks
> 
> -- 
> Toqeer Ali Syed
> 
> Red Hat Certified Engineer
> mob:     +92 321 9059916
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From talk2ram at gmail.com  Fri May 14 09:11:07 2010
From: talk2ram at gmail.com (ram)
Date: Fri, 14 May 2010 21:41:07 +0530
Subject: [Freeswitch-users] freeswitch RPM available for centos 5.4
In-Reply-To: <9856D01D-A4FD-4C6F-8E01-48426E7C0557@freeswitch.org>
References: <AANLkTinzMmrg3BqZHmpA7TeBPR_0jEYRchGuXr9dNEDZ@mail.gmail.com>
	<9856D01D-A4FD-4C6F-8E01-48426E7C0557@freeswitch.org>
Message-ID: <AANLkTimcIozBOpd4XC8iiDuwVQZ9PWA7bXiBCs0L4uFZ@mail.gmail.com>

On Fri, May 14, 2010 at 8:24 PM, Brian West <brian at freeswitch.org> wrote:

> While I welcome your RPM's you should at the very least compile 64bit
> version too... we are currently working to provide, 32,64 for CentOS, Debian
> and Ubuntu along with nightly builds of OS X and Windows 32bit ... If anyone
> wishes to help configure a host in our build farm we are going to deploy
> later this month please email me.
>
> Thanks,
> Brian
>
>

will soon come with 64bit too



>    On May 14, 2010, at 9:46 AM, ram wrote:
>
>   hi all
>
> we have recently build freeswitch RPM  ( based on 13th may 2010 git) for
> fusionpbx ISO
>
> you can try installing ISO or RPMS
>
> http://wiki.fusionpbx.com/index.php/CentOS_ISO
>
>
> Ram
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jerry.richards at teotech.com  Fri May 14 10:15:01 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 14 May 2010 10:15:01 -0700
Subject: [Freeswitch-users] Auto-Callback For Valet Parked Call
Message-ID: <1D4812CDCBB046DF8AAD389139EA8C37@greyhawk.tonecommander.com>


I like the valet parking feature.  Is there a tag for
calling-back/re-connecting the one who parked the call, if the parked call
sits there too long (e.g. more than 2 minutes)?

Thanks And Best Regards,
Jerry 



From moises.silva at gmail.com  Fri May 14 10:35:20 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Fri, 14 May 2010 13:35:20 -0400
Subject: [Freeswitch-users] mod_openzap, sangoma, FAX and HW-HEC
In-Reply-To: <4BED4E95.8090508@coppice.org>
References: <4BEADBBC.5090802@ewetel.de>
	<AANLkTiksacBjB7uuCzZidLX5PxVFmNsE7FHNskSVjhnM@mail.gmail.com>
	<AANLkTimVaDRo7ypzZu4YmgeVVMdV-sQ4TEocJOHaFv-F@mail.gmail.com>
	<AANLkTikUPqLyeeToYSj9SeJEdkz5cLsMT6_r6ZjEJyc3@mail.gmail.com>
	<4BED4E95.8090508@coppice.org>
Message-ID: <AANLkTikX1C2Iltl-u4mKDOxanWJCKG7E_Wmzrj2k_MTi@mail.gmail.com>

No I don't have any idea of what the octasic chip does exactly when that
setting is enabled. Nenad disabled that setting by default starting with
driver 3.5.11 since indeed was related to many fax problems.

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com


On Fri, May 14, 2010 at 9:22 AM, Steve Underwood <steveu at coppice.org> wrote:

> Hi Moises,
>
> Do you know exactly what the noise reduction does? In an echo canceller
> context I can see a couple of possibilities. I think all of them are
> pretty horrible for what they might do to modem signals. Maybe this has
> something to do with why so many people have problems with both Sangoma
> and Digium boards that use the Octasic echo cancellers.
>
> Steve
>
>
> On 05/13/2010 03:47 AM, Moises Silva wrote:
> > Sorry, the setting should be under [wanpipex] in wanpipex.conf file
> > and NOT in the [wxgx] section.
> >
> > Moises Silva
> > Senior Software Engineer
> > Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON
> > L3R 9T3 Canada
> > t. 1 905 474 1990 x 128 | e. moy at sangoma.com <mailto:moy at sangoma.com>
> >
> >
> > On Wed, May 12, 2010 at 3:40 PM, Moises Silva <moises.silva at gmail.com
> > <mailto:moises.silva at gmail.com>> wrote:
> >
> >     It is implemented and should work as a work-around.
> >
> >     Having said that, I suggest you to contact Sangoma support
> >     (techdesk at sangoma.com <mailto:techdesk at sangoma.com>), the ec chip
> >     should detect fax tone and disable its echo-canceling functions.
> >     However, some old versions of the driver have a noise-reduction
> >     feature enabled by default which also happen to disrupt faxes.
> >
> >     Try adding this just below TDMV_HWEC=YES option:
> >
> >     HWEC_NOISE_REDUCTION_DISABLE=YES
> >
> >     Or upgrade to the latest driver which does this automatically.
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From moises.silva at gmail.com  Fri May 14 10:36:10 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Fri, 14 May 2010 13:36:10 -0400
Subject: [Freeswitch-users] mod_openzap, sangoma, FAX and HW-HEC
In-Reply-To: <4BECE9CC.9080202@ewetel.de>
References: <4BEADBBC.5090802@ewetel.de>
	<AANLkTiksacBjB7uuCzZidLX5PxVFmNsE7FHNskSVjhnM@mail.gmail.com>
	<AANLkTimVaDRo7ypzZu4YmgeVVMdV-sQ4TEocJOHaFv-F@mail.gmail.com>
	<AANLkTikUPqLyeeToYSj9SeJEdkz5cLsMT6_r6ZjEJyc3@mail.gmail.com>
	<4BECE9CC.9080202@ewetel.de>
Message-ID: <AANLkTimzhAlrWFftE34J5oyPp4KbiZmhEvCUaAlzyo3B@mail.gmail.com>

3.5.10 is not an old version, but the noise reduction is disabled by default
starting with 3.5.11 according to the change log.

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com


On Fri, May 14, 2010 at 2:12 AM, Helmut Kuper <helmut.kuper at ewetel.de>wrote:

> Hello Moises,
>
> thx. Well I saw a hint for noise reduction capabilities of A104d
> somewhere in the logs during debugging. And it's easy to suppose that
> fax data could be interpreted as noise. On the other hand I'm a bit
> confused because the Fax is transmitted in a digital way, so where
> should the noise comes from (except the quantization noise).
>
>
> Oh, and I use wanpipe "WANPIPE Release: 3.5.10". Is this version already
> too old?
>
> I will try to activated this new work around to disable HEC per channel
> on demand.
>
> Thx alot to both of you.
>
> regards
> helmut
>
>
>
> On 12.05.2010 21:47, Moises Silva wrote:
> > Sorry, the setting should be under [wanpipex] in wanpipex.conf file and
> > NOT in the [wxgx] section.
> >
> > Moises Silva
> > Senior Software Engineer
> > Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
> > 9T3 Canada
> > t. 1 905 474 1990 x 128 | e. moy at sangoma.com <mailto:moy at sangoma.com>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jerry.richards at teotech.com  Fri May 14 11:08:11 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 14 May 2010 11:08:11 -0700
Subject: [Freeswitch-users] Core Dump
Message-ID: <3FFC6E3833164D8784DB28E73079194F@greyhawk.tonecommander.com>

Our Freeswitch server crashed over night and there is a core dump file (it
is a snapshot of the trunk that existed on 04/27/10).  Are you interested in
seeing the core dump?  If so, does the pastebin support that type of file?

Thanks,
Jerry



From rupa at rupa.com  Fri May 14 11:20:23 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 14 May 2010 13:20:23 -0500
Subject: [Freeswitch-users] Core Dump
In-Reply-To: <3FFC6E3833164D8784DB28E73079194F@greyhawk.tonecommander.com>
References: <3FFC6E3833164D8784DB28E73079194F@greyhawk.tonecommander.com>
Message-ID: <AANLkTim1DXFDDSuH3Pwu9dMwoYLCmXaMWHPZgV2_LMp8@mail.gmail.com>

There is handy-dandy script in support-d called fscore_pb.  Run it from the
directory in which the core dump is in:

/path/to/fscore_pb name

where name is the name you want to show up on the pastebin.

Usually we ask that bug reports go through jira (include the link to the
pastebin that fscore_pb gives you)

On Fri, May 14, 2010 at 1:08 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

> Our Freeswitch server crashed over night and there is a core dump file (it
> is a snapshot of the trunk that existed on 04/27/10).  Are you interested
> in
> seeing the core dump?  If so, does the pastebin support that type of file?
>
> Thanks,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
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From mike at jerris.com  Fri May 14 11:29:30 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 14 May 2010 14:29:30 -0400
Subject: [Freeswitch-users] Core Dump
In-Reply-To: <3FFC6E3833164D8784DB28E73079194F@greyhawk.tonecommander.com>
References: <3FFC6E3833164D8784DB28E73079194F@greyhawk.tonecommander.com>
Message-ID: <BAE0B892-C269-4C1C-AFD3-735345F89DD7@jerris.com>

If you had a segfault please post a backtrace of this to our bugtracker at http://jira.freeswitch.org

http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#Simple_bash_script_to_make_debug_easy
http://wiki.freeswitch.org/wiki/Reporting_Bugs



On May 14, 2010, at 2:08 PM, Jerry Richards wrote:

> Our Freeswitch server crashed over night and there is a core dump file (it
> is a snapshot of the trunk that existed on 04/27/10).  Are you interested in
> seeing the core dump?  If so, does the pastebin support that type of file?



From jerry.richards at teotech.com  Fri May 14 11:56:11 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Fri, 14 May 2010 11:56:11 -0700
Subject: [Freeswitch-users] Core Dump
In-Reply-To: <AANLkTim1DXFDDSuH3Pwu9dMwoYLCmXaMWHPZgV2_LMp8@mail.gmail.com>
References: <3FFC6E3833164D8784DB28E73079194F@greyhawk.tonecommander.com>
	<AANLkTim1DXFDDSuH3Pwu9dMwoYLCmXaMWHPZgV2_LMp8@mail.gmail.com>
Message-ID: <B4F5368326594C2FB2893C86E3A3FD1B@greyhawk.tonecommander.com>

When I run support-d/fscore_pb, I get the following output (and it didn't
put anything into the pastebin):
 
[root at TeoProxy Freeswitch]# support-d/fscore_pb core.26916
Found gdb
core is core.26916
Gathering Data Please Wait.........
ls: [0-9]*: No such file or directory
Finished.
Please report  <http://pastebin.freeswitch.org/>
http://pastebin.freeswitch.org/ to the developers.
[root at TeoProxy Freeswitch]#

Jerry


  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa
Schomaker
Sent: Friday, May 14, 2010 11:20 AM
To: freeswitch-users
Subject: Re: [Freeswitch-users] Core Dump


There is handy-dandy script in support-d called fscore_pb.  Run it from the
directory in which the core dump is in: 

/path/to/fscore_pb name

where name is the name you want to show up on the pastebin.  

Usually we ask that bug reports go through jira (include the link to the
pastebin that fscore_pb gives you)


On Fri, May 14, 2010 at 1:08 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:


Our Freeswitch server crashed over night and there is a core dump file (it
is a snapshot of the trunk that existed on 04/27/10).  Are you interested in
seeing the core dump?  If so, does the pastebin support that type of file?

Thanks,
Jerry


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





-- 
-Rupa


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From ron.freeswitch at mcleodnet.com  Fri May 14 12:21:01 2010
From: ron.freeswitch at mcleodnet.com (Ron McLeod)
Date: Fri, 14 May 2010 12:21:01 -0700
Subject: [Freeswitch-users] Core Dump
In-Reply-To: <B4F5368326594C2FB2893C86E3A3FD1B@greyhawk.tonecommander.com>
References: <3FFC6E3833164D8784DB28E73079194F@greyhawk.tonecommander.com><AANLkTim1DXFDDSuH3Pwu9dMwoYLCmXaMWHPZgV2_LMp8@mail.gmail.com>
	<B4F5368326594C2FB2893C86E3A3FD1B@greyhawk.tonecommander.com>
Message-ID: <C7FD3B194CAD42BBBADBF2899B379DD9@fromage>

Did you  "Run it from the directory in which the core dump is in" ?

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jerry
Richards
Sent: Friday, May 14, 2010 11:56 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Core Dump

 

When I run support-d/fscore_pb, I get the following output (and it didn't
put anything into the pastebin):

 

[root at TeoProxy Freeswitch]# support-d/fscore_pb core.26916
Found gdb
core is core.26916
Gathering Data Please Wait.........
ls: [0-9]*: No such file or directory
Finished.
Please report  <http://pastebin.freeswitch.org/>
http://pastebin.freeswitch.org/ to the developers.
[root at TeoProxy Freeswitch]#

Jerry

 


  _____  


From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa
Schomaker
Sent: Friday, May 14, 2010 11:20 AM
To: freeswitch-users
Subject: Re: [Freeswitch-users] Core Dump

There is handy-dandy script in support-d called fscore_pb.  Run it from the
directory in which the core dump is in: 

 

/path/to/fscore_pb name

 

where name is the name you want to show up on the pastebin.  

 

Usually we ask that bug reports go through jira (include the link to the
pastebin that fscore_pb gives you)

On Fri, May 14, 2010 at 1:08 PM, Jerry Richards <jerry.richards at teotech.com>
wrote:

Our Freeswitch server crashed over night and there is a core dump file (it
is a snapshot of the trunk that existed on 04/27/10).  Are you interested in
seeing the core dump?  If so, does the pastebin support that type of file?

Thanks,
Jerry


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




-- 
-Rupa

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From rupa at rupa.com  Fri May 14 12:21:35 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 14 May 2010 14:21:35 -0500
Subject: [Freeswitch-users] Core Dump
In-Reply-To: <B4F5368326594C2FB2893C86E3A3FD1B@greyhawk.tonecommander.com>
References: <3FFC6E3833164D8784DB28E73079194F@greyhawk.tonecommander.com>
	<AANLkTim1DXFDDSuH3Pwu9dMwoYLCmXaMWHPZgV2_LMp8@mail.gmail.com>
	<B4F5368326594C2FB2893C86E3A3FD1B@greyhawk.tonecommander.com>
Message-ID: <AANLkTilN5fCpsrt3wdYwm4feajygHAAIgGy_UfGlN_XB@mail.gmail.com>

Ok, well, do it manually like the wiki says.  Do you irc?  I'd like to
understand why fscore_pb is failing for you but it would probably require
some interaction to address.  Unless you are familiar with shell scripting
and can identify the failure point.

On Fri, May 14, 2010 at 1:56 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

>  When I run support-d/fscore_pb, I get the following output (and it didn't
> put anything into the pastebin):
>
> [root at TeoProxy Freeswitch]# support-d/fscore_pb core.26916
> Found gdb
> core is core.26916
> Gathering Data Please Wait.........
> ls: [0-9]*: No such file or directory
> Finished.
> Please report http://pastebin.freeswitch.org/ to the developers.
> [root at TeoProxy Freeswitch]#
> Jerry
>
>  ------------------------------
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Rupa
> Schomaker
> *Sent:* Friday, May 14, 2010 11:20 AM
> *To:* freeswitch-users
> *Subject:* Re: [Freeswitch-users] Core Dump
>
> There is handy-dandy script in support-d called fscore_pb.  Run it from the
> directory in which the core dump is in:
>
> /path/to/fscore_pb name
>
> where name is the name you want to show up on the pastebin.
>
> Usually we ask that bug reports go through jira (include the link to the
> pastebin that fscore_pb gives you)
>
> On Fri, May 14, 2010 at 1:08 PM, Jerry Richards <
> jerry.richards at teotech.com> wrote:
>
>> Our Freeswitch server crashed over night and there is a core dump file (it
>> is a snapshot of the trunk that existed on 04/27/10).  Are you interested
>> in
>> seeing the core dump?  If so, does the pastebin support that type of file?
>>
>> Thanks,
>> Jerry
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> -Rupa
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From rupa at rupa.com  Fri May 14 12:34:11 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 14 May 2010 14:34:11 -0500
Subject: [Freeswitch-users] Core Dump
In-Reply-To: <C7FD3B194CAD42BBBADBF2899B379DD9@fromage>
References: <3FFC6E3833164D8784DB28E73079194F@greyhawk.tonecommander.com>
	<AANLkTim1DXFDDSuH3Pwu9dMwoYLCmXaMWHPZgV2_LMp8@mail.gmail.com>
	<B4F5368326594C2FB2893C86E3A3FD1B@greyhawk.tonecommander.com>
	<C7FD3B194CAD42BBBADBF2899B379DD9@fromage>
Message-ID: <AANLkTik2-Qo8GRSVNDCxQkL1Iy-hPVjgu8bWNhtRTjLw@mail.gmail.com>

Yeah, he did -- it identified the core file.  It is borking up later.

On Fri, May 14, 2010 at 2:21 PM, Ron McLeod <ron.freeswitch at mcleodnet.com>wrote:

>  Did you  ?Run it from the directory in which the core dump is in? ?
>
>
>   ------------------------------
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Jerry
> Richards
> *Sent:* Friday, May 14, 2010 11:56 AM
> *To:* freeswitch-users at lists.freeswitch.org
>
> *Subject:* Re: [Freeswitch-users] Core Dump
>
>
>
> When I run support-d/fscore_pb, I get the following output (and it didn't
> put anything into the pastebin):
>
>
>
> [root at TeoProxy Freeswitch]# support-d/fscore_pb core.26916
> Found gdb
> core is core.26916
> Gathering Data Please Wait.........
> ls: [0-9]*: No such file or directory
> Finished.
> Please report http://pastebin.freeswitch.org/ to the developers.
> [root at TeoProxy Freeswitch]#
>
> Jerry
>
>
>  ------------------------------
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Rupa
> Schomaker
> *Sent:* Friday, May 14, 2010 11:20 AM
> *To:* freeswitch-users
> *Subject:* Re: [Freeswitch-users] Core Dump
>
> There is handy-dandy script in support-d called fscore_pb.  Run it from the
> directory in which the core dump is in:
>
>
>
> /path/to/fscore_pb name
>
>
>
> where name is the name you want to show up on the pastebin.
>
>
>
> Usually we ask that bug reports go through jira (include the link to the
> pastebin that fscore_pb gives you)
>
> On Fri, May 14, 2010 at 1:08 PM, Jerry Richards <
> jerry.richards at teotech.com> wrote:
>
> Our Freeswitch server crashed over night and there is a core dump file (it
> is a snapshot of the trunk that existed on 04/27/10).  Are you interested
> in
> seeing the core dump?  If so, does the pastebin support that type of file?
>
> Thanks,
> Jerry
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> -Rupa
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From toqeer83 at gmail.com  Fri May 14 12:39:40 2010
From: toqeer83 at gmail.com (toqeer ali)
Date: Fri, 14 May 2010 12:39:40 -0700
Subject: [Freeswitch-users] Hunting
In-Reply-To: <4BED7112.1040701@communicatefreely.net>
References: <AANLkTinW51XkJQAqDdI-D8760rrVOnMfaFomGjzLH7-L@mail.gmail.com>
	<4BED7112.1040701@communicatefreely.net>
Message-ID: <AANLkTikEr7gTTHWvYVnNTPhu7R6FzCohsg14pMSUE_UR@mail.gmail.com>

Hi Tim St. Pierr

Thanks you very much for your reply i got the point.

But one question more please, can i also define the time for example for 60
second one extension could not answer go to other one?

Thanks

On Fri, May 14, 2010 at 8:49 AM, Tim St. Pierre <
fs-list at communicatefreely.net> wrote:

> Nope, no special module required.
>
> Before you call the bridge application, set continue_on_fail=true
> This will tell freeswitch to keep going in the dialplan if the first
> destination doesn't answer.
> After the first bridge application, add other entries that take try the
> next destination.  You could
> use the transfer app to try another extension as you described.
>
> -Tim
>
> toqeer ali wrote:
> > Hi all,
> >
> > I am new to freeswitch and want to implement Hunting.
> >
> > For example if call come to my freeswitch box where DID is configured to
> > an extension for example 1000 so if 1000 did not pick the phone then it
> > should be transfered to other extension or landline.
> >
> > Please give me a clue... Is there is any special module for that?
> >
> > Any help will highly appreciated.
> >
> > Thanks
> >
> > --
> > Toqeer Ali Syed
> >
> > Red Hat Certified Engineer
> > mob:     +92 321 9059916
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Toqeer Ali Syed

Red Hat Certified Engineer
mob:     +92 321 9059916
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From steve.d.ward at gmail.com  Fri May 14 13:46:37 2010
From: steve.d.ward at gmail.com (Steven Ward)
Date: Fri, 14 May 2010 13:46:37 -0700
Subject: [Freeswitch-users] Hunting
In-Reply-To: <AANLkTikEr7gTTHWvYVnNTPhu7R6FzCohsg14pMSUE_UR@mail.gmail.com>
References: <AANLkTinW51XkJQAqDdI-D8760rrVOnMfaFomGjzLH7-L@mail.gmail.com>
	<4BED7112.1040701@communicatefreely.net>
	<AANLkTikEr7gTTHWvYVnNTPhu7R6FzCohsg14pMSUE_UR@mail.gmail.com>
Message-ID: <AANLkTimJhhBIxFOuMcWjaa9SHnwAvQ7yvUzOl5e_YrJC@mail.gmail.com>

As for timing out a bridge attempt...

Before you do the bridge, one option is to set the call_timeout channel
variable to 60.

But, if the destination endpoint sends early media, you'll also have to set
ignore_early_media to true before you do the bridge, since call_timeout only
applies until the call attempt is successful (it's successful when you
either get early media or a true answer).

You can check out
http://wiki.freeswitch.org/wiki/Channel_Variables#Timeout_Related

Of course, it's all about the details of what you're doing and why, but
those options should provide a really good place to start.


On Fri, May 14, 2010 at 12:39 PM, toqeer ali <toqeer83 at gmail.com> wrote:

> Hi Tim St. Pierr
>
> Thanks you very much for your reply i got the point.
>
> But one question more please, can i also define the time for example for 60
> second one extension could not answer go to other one?
>
> Thanks
>
> On Fri, May 14, 2010 at 8:49 AM, Tim St. Pierre <
> fs-list at communicatefreely.net> wrote:
>
>> Nope, no special module required.
>>
>> Before you call the bridge application, set continue_on_fail=true
>> This will tell freeswitch to keep going in the dialplan if the first
>> destination doesn't answer.
>> After the first bridge application, add other entries that take try the
>> next destination.  You could
>> use the transfer app to try another extension as you described.
>>
>> -Tim
>>
>> toqeer ali wrote:
>> > Hi all,
>> >
>> > I am new to freeswitch and want to implement Hunting.
>> >
>> > For example if call come to my freeswitch box where DID is configured to
>> > an extension for example 1000 so if 1000 did not pick the phone then it
>> > should be transfered to other extension or landline.
>> >
>> > Please give me a clue... Is there is any special module for that?
>> >
>> > Any help will highly appreciated.
>> >
>> > Thanks
>> >
>> > --
>> > Toqeer Ali Syed
>> >
>> > Red Hat Certified Engineer
>> > mob:     +92 321 9059916
>> >
>> >
>> > ------------------------------------------------------------------------
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Toqeer Ali Syed
>
> Red Hat Certified Engineer
> mob:     +92 321 9059916
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jan.berger at video24.no  Fri May 14 13:52:50 2010
From: jan.berger at video24.no (Jan Berger)
Date: Fri, 14 May 2010 22:52:50 +0200
Subject: [Freeswitch-users] Skype SIP
Message-ID: <F42185CF9FCC458AB72C98D8D2AEE9D0@dell9400>

Hi,

 

Do anyone have experience using the Skype SIP trunking?

 

Is this standard SIP and does it connect smoothly to FreeSWITCH?

 

Jan

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From brian at freeswitch.org  Fri May 14 13:58:55 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 14 May 2010 15:58:55 -0500
Subject: [Freeswitch-users] Skype SIP
In-Reply-To: <F42185CF9FCC458AB72C98D8D2AEE9D0@dell9400>
References: <F42185CF9FCC458AB72C98D8D2AEE9D0@dell9400>
Message-ID: <3EA5BADA-6F71-45C0-85A5-FC3CB9CAC778@freeswitch.org>

yes and yes.

/b

On May 14, 2010, at 3:52 PM, Jan Berger wrote:

> Is this standard SIP and does it connect smoothly to FreeSWITCH?

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From jan.berger at video24.no  Fri May 14 14:10:48 2010
From: jan.berger at video24.no (Jan Berger)
Date: Fri, 14 May 2010 23:10:48 +0200
Subject: [Freeswitch-users] Skype SIP
In-Reply-To: <3EA5BADA-6F71-45C0-85A5-FC3CB9CAC778@freeswitch.org>
References: <F42185CF9FCC458AB72C98D8D2AEE9D0@dell9400>
	<3EA5BADA-6F71-45C0-85A5-FC3CB9CAC778@freeswitch.org>
Message-ID: <7D8EBB3890614E4F99D016C27562810D@dell9400>

We better get Skype certification then :-) - or do we have that as well?

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: 14. mai 2010 22:59
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Skype SIP

 

yes and yes.

 

/b

 

On May 14, 2010, at 3:52 PM, Jan Berger wrote:





Is this standard SIP and does it connect smoothly to FreeSWITCH?

 

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From brian at freeswitch.org  Fri May 14 14:15:02 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 14 May 2010 16:15:02 -0500
Subject: [Freeswitch-users] Skype SIP
In-Reply-To: <7D8EBB3890614E4F99D016C27562810D@dell9400>
References: <F42185CF9FCC458AB72C98D8D2AEE9D0@dell9400>
	<3EA5BADA-6F71-45C0-85A5-FC3CB9CAC778@freeswitch.org>
	<7D8EBB3890614E4F99D016C27562810D@dell9400>
Message-ID: <A40DAAD1-D8AB-4AFA-A6F4-B7781585B2D4@freeswitch.org>

Why do you need a Skype certification?

/b

On May 14, 2010, at 4:10 PM, Jan Berger wrote:

> We better get Skype certification then J - or do we have that as well?
>  

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From msc at freeswitch.org  Fri May 14 14:35:42 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 14 May 2010 14:35:42 -0700
Subject: [Freeswitch-users] Call for help: stuff to document
Message-ID: <AANLkTikDy7qxSsSzIbNUNsm-ssPnr_5sez3WEs8HcG7C@mail.gmail.com>

Hello all,

The FS devs have been busy adding features (as usual) and we need help
getting them all documented (as usual). I've added them all to next week's
agenda page:

http://wiki.freeswitch.org/wiki/FS_weekly_2010_05_19

There are a lot of new things listed under "Items needing documentation" -
if you are able to document all of these please feel free to do so. Be sure
to update the page by linking to any documentation you create.

Thanks!

-Michael
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From jan.berger at video24.no  Fri May 14 15:42:57 2010
From: jan.berger at video24.no (Jan Berger)
Date: Sat, 15 May 2010 00:42:57 +0200
Subject: [Freeswitch-users] Skype SIP
In-Reply-To: <A40DAAD1-D8AB-4AFA-A6F4-B7781585B2D4@freeswitch.org>
References: <F42185CF9FCC458AB72C98D8D2AEE9D0@dell9400>
	<3EA5BADA-6F71-45C0-85A5-FC3CB9CAC778@freeswitch.org>
	<7D8EBB3890614E4F99D016C27562810D@dell9400>
	<A40DAAD1-D8AB-4AFA-A6F4-B7781585B2D4@freeswitch.org>
Message-ID: <D357454F9B4044818470854B86C86DAA@dell9400>

Skype maintain a list of vendors who deliver "approved" equipment that is
compatible with skype - just nice if we get on that list.

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: 14. mai 2010 23:15
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Skype SIP

 

Why do you need a Skype certification?

 

/b

 

On May 14, 2010, at 4:10 PM, Jan Berger wrote:





We better get Skype certification then :-) - or do we have that as well?

 

 

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From freeswitch at gilligan.id.au  Fri May 14 17:52:56 2010
From: freeswitch at gilligan.id.au (Chris)
Date: Sat, 15 May 2010 10:52:56 +1000
Subject: [Freeswitch-users] mod_managed or a script,
	sending DTMF and 	receiving DTMF
In-Reply-To: <AANLkTimDZx2T15GHnWck36bESESuK8n4u_wzzujyxvfE@mail.gmail.com>
References: <AANLkTikSTwOw1yiBGO9VrVDnsKtH7ySz8V2LxL1XUdru@mail.gmail.com> 
	<AANLkTik1BYHsuHlfS9FAD3pEGS5GaEIINiiY9708sFji@mail.gmail.com> 
	<AANLkTimDZx2T15GHnWck36bESESuK8n4u_wzzujyxvfE@mail.gmail.com>
Message-ID: <AANLkTilXbUSK5DbDb5RWboqEJLB6xAuPVwnDA21KsuZK@mail.gmail.com>

sadly uuid_send_dtmf did not seem to work


On Fri, May 14, 2010 at 11:10 AM, Chris <freeswitch at gilligan.id.au> wrote:

> Thanks so that is the only way?  I will have to do some tests and see if i
> can capture dtmf at the same time as i send when i do it that way.  Any
> other options?
>
>
> On Thu, May 13, 2010 at 10:50 PM, Phillip Jones <pjintheusa at gmail.com>wrote:
>
>> Try using http://wiki.freeswitch.org/wiki/Mod_commands#uuid_send_dtmf
>>
>> On Thu, May 13, 2010 at 5:32 AM, Chris <freeswitch at gilligan.id.au> wrote:
>>
>>> HI,
>>> I am currently working on a project in mod_managed and i am trying to
>>> discover the best way to meet my requirements.  i am hoping someone will
>>> have some ideas.  This could be implemented in one of the other scripting
>>> language if needed.
>>>
>>> What i am trying to do is reprogram a remote device via the phone.  The
>>> device takes commands in the form of DTMF tones and responds in different
>>> DTMF tones depending on success or failure. An example of the flow would be
>>>
>>> freeswitch -> 342523#
>>> device -> 1
>>> freeswitch -> 356789#
>>> device ->2
>>>
>>> device always responds with one digit and freeswitch sends many.
>>>
>>> I was hoping to do this with a database and mod_managed but i can't
>>> workout how to send the DTMF in mod_managed unless i user audio files for it
>>> which seems to be the wrong way to go to me. While i would prefer a
>>> mod_managed solution i will take anything i can find.
>>>
>>>
>>>
>>> Chris
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From infos at madovsky.org  Fri May 14 18:22:47 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 14 May 2010 21:22:47 -0400
Subject: [Freeswitch-users] mod_managed or a script,
	sending DTMF and 	receiving DTMF
References: <AANLkTikSTwOw1yiBGO9VrVDnsKtH7ySz8V2LxL1XUdru@mail.gmail.com>
	<AANLkTik1BYHsuHlfS9FAD3pEGS5GaEIINiiY9708sFji@mail.gmail.com>
	<AANLkTimDZx2T15GHnWck36bESESuK8n4u_wzzujyxvfE@mail.gmail.com>
	<AANLkTilXbUSK5DbDb5RWboqEJLB6xAuPVwnDA21KsuZK@mail.gmail.com>
Message-ID: <B429D30727D34D35B0FA648556D337F3@MOBILEE1705>

it works, I use it everyday
  ----- Original Message ----- 
  From: Chris 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, May 14, 2010 8:52 PM
  Subject: Re: [Freeswitch-users] mod_managed or a script,sending DTMF and receiving DTMF


  sadly uuid_send_dtmf did not seem to work



  On Fri, May 14, 2010 at 11:10 AM, Chris <freeswitch at gilligan.id.au> wrote:

    Thanks so that is the only way?  I will have to do some tests and see if i can capture dtmf at the same time as i send when i do it that way.  Any other options?



    On Thu, May 13, 2010 at 10:50 PM, Phillip Jones <pjintheusa at gmail.com> wrote:

      Try using http://wiki.freeswitch.org/wiki/Mod_commands#uuid_send_dtmf


      On Thu, May 13, 2010 at 5:32 AM, Chris <freeswitch at gilligan.id.au> wrote:

        HI,
        I am currently working on a project in mod_managed and i am trying to discover the best way to meet my requirements.  i am hoping someone will have some ideas.  This could be implemented in one of the other scripting language if needed.


        What i am trying to do is reprogram a remote device via the phone.  The device takes commands in the form of DTMF tones and responds in different DTMF tones depending on success or failure. An example of the flow would be


        freeswitch -> 342523#
        device -> 1
        freeswitch -> 356789#
        device ->2


        device always responds with one digit and freeswitch sends many.


        I was hoping to do this with a database and mod_managed but i can't workout how to send the DTMF in mod_managed unless i user audio files for it which seems to be the wrong way to go to me. While i would prefer a mod_managed solution i will take anything i can find.






        Chris




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




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








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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From freeswitch at gilligan.id.au  Fri May 14 18:33:29 2010
From: freeswitch at gilligan.id.au (Chris)
Date: Sat, 15 May 2010 11:33:29 +1000
Subject: [Freeswitch-users] mod_managed or a script,
	sending DTMF and 	receiving DTMF
In-Reply-To: <B429D30727D34D35B0FA648556D337F3@MOBILEE1705>
References: <AANLkTikSTwOw1yiBGO9VrVDnsKtH7ySz8V2LxL1XUdru@mail.gmail.com> 
	<AANLkTik1BYHsuHlfS9FAD3pEGS5GaEIINiiY9708sFji@mail.gmail.com> 
	<AANLkTimDZx2T15GHnWck36bESESuK8n4u_wzzujyxvfE@mail.gmail.com> 
	<AANLkTilXbUSK5DbDb5RWboqEJLB6xAuPVwnDA21KsuZK@mail.gmail.com> 
	<B429D30727D34D35B0FA648556D337F3@MOBILEE1705>
Message-ID: <AANLkTimGSAW5M9dmu2r70M2AM6jRMcSWZxjA__K7h8XW@mail.gmail.com>

do you happen to be using ti from inside mod_managed or via a different
means?  Any chance of a small sample?  maybe i am doing something stupid.

On Sat, May 15, 2010 at 11:22 AM, Madovsky <infos at madovsky.org> wrote:

>  it works, I use it everyday
>
> ----- Original Message -----
> *From:* Chris <freeswitch at gilligan.id.au>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Friday, May 14, 2010 8:52 PM
> *Subject:* Re: [Freeswitch-users] mod_managed or a script,sending DTMF and
> receiving DTMF
>
> sadly uuid_send_dtmf did not seem to work
>
>
> On Fri, May 14, 2010 at 11:10 AM, Chris <freeswitch at gilligan.id.au> wrote:
>
>> Thanks so that is the only way?  I will have to do some tests and see if i
>> can capture dtmf at the same time as i send when i do it that way.  Any
>> other options?
>>
>>
>> On Thu, May 13, 2010 at 10:50 PM, Phillip Jones <pjintheusa at gmail.com>wrote:
>>
>>> Try using http://wiki.freeswitch.org/wiki/Mod_commands#uuid_send_dtmf
>>>
>>>   On Thu, May 13, 2010 at 5:32 AM, Chris <freeswitch at gilligan.id.au>wrote:
>>>
>>>>  HI,
>>>> I am currently working on a project in mod_managed and i am trying to
>>>> discover the best way to meet my requirements.  i am hoping someone will
>>>> have some ideas.  This could be implemented in one of the other scripting
>>>> language if needed.
>>>>
>>>> What i am trying to do is reprogram a remote device via the phone.  The
>>>> device takes commands in the form of DTMF tones and responds in different
>>>> DTMF tones depending on success or failure. An example of the flow would be
>>>>
>>>> freeswitch -> 342523#
>>>> device -> 1
>>>> freeswitch -> 356789#
>>>> device ->2
>>>>
>>>> device always responds with one digit and freeswitch sends many.
>>>>
>>>> I was hoping to do this with a database and mod_managed but i can't
>>>> workout how to send the DTMF in mod_managed unless i user audio files for it
>>>> which seems to be the wrong way to go to me. While i would prefer a
>>>> mod_managed solution i will take anything i can find.
>>>>
>>>>
>>>>
>>>> Chris
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Fri May 14 18:39:13 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 14 May 2010 21:39:13 -0400
Subject: [Freeswitch-users] mod_managed or a script,
	sending DTMF and 	receiving DTMF
References: <AANLkTikSTwOw1yiBGO9VrVDnsKtH7ySz8V2LxL1XUdru@mail.gmail.com>
	<AANLkTik1BYHsuHlfS9FAD3pEGS5GaEIINiiY9708sFji@mail.gmail.com>
	<AANLkTimDZx2T15GHnWck36bESESuK8n4u_wzzujyxvfE@mail.gmail.com>
	<AANLkTilXbUSK5DbDb5RWboqEJLB6xAuPVwnDA21KsuZK@mail.gmail.com>
	<B429D30727D34D35B0FA648556D337F3@MOBILEE1705>
	<AANLkTimGSAW5M9dmu2r70M2AM6jRMcSWZxjA__K7h8XW@mail.gmail.com>
Message-ID: <7350123A2AAC421587893A6BD6657C74@MOBILEE1705>

I use it with fs_cli
  ----- Original Message ----- 
  From: Chris 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, May 14, 2010 9:33 PM
  Subject: Re: [Freeswitch-users] mod_managed or a script,sending DTMF and receiving DTMF


  do you happen to be using ti from inside mod_managed or via a different means?  Any chance of a small sample?  maybe i am doing something stupid.


  On Sat, May 15, 2010 at 11:22 AM, Madovsky <infos at madovsky.org> wrote:

    it works, I use it everyday
      ----- Original Message ----- 
      From: Chris 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Friday, May 14, 2010 8:52 PM
      Subject: Re: [Freeswitch-users] mod_managed or a script,sending DTMF and receiving DTMF


      sadly uuid_send_dtmf did not seem to work 



      On Fri, May 14, 2010 at 11:10 AM, Chris <freeswitch at gilligan.id.au> wrote:

        Thanks so that is the only way?  I will have to do some tests and see if i can capture dtmf at the same time as i send when i do it that way.  Any other options? 



        On Thu, May 13, 2010 at 10:50 PM, Phillip Jones <pjintheusa at gmail.com> wrote:

          Try using http://wiki.freeswitch.org/wiki/Mod_commands#uuid_send_dtmf


          On Thu, May 13, 2010 at 5:32 AM, Chris <freeswitch at gilligan.id.au> wrote:

            HI, 
            I am currently working on a project in mod_managed and i am trying to discover the best way to meet my requirements.  i am hoping someone will have some ideas.  This could be implemented in one of the other scripting language if needed.


            What i am trying to do is reprogram a remote device via the phone.  The device takes commands in the form of DTMF tones and responds in different DTMF tones depending on success or failure. An example of the flow would be


            freeswitch -> 342523#
            device -> 1
            freeswitch -> 356789#
            device ->2


            device always responds with one digit and freeswitch sends many.


            I was hoping to do this with a database and mod_managed but i can't workout how to send the DTMF in mod_managed unless i user audio files for it which seems to be the wrong way to go to me. While i would prefer a mod_managed solution i will take anything i can find.






            Chris




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




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








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


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



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






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


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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From freeswitch at gilligan.id.au  Fri May 14 18:45:47 2010
From: freeswitch at gilligan.id.au (Chris)
Date: Sat, 15 May 2010 11:45:47 +1000
Subject: [Freeswitch-users] mod_managed or a script,
	sending DTMF and 	receiving DTMF
In-Reply-To: <7350123A2AAC421587893A6BD6657C74@MOBILEE1705>
References: <AANLkTikSTwOw1yiBGO9VrVDnsKtH7ySz8V2LxL1XUdru@mail.gmail.com> 
	<AANLkTik1BYHsuHlfS9FAD3pEGS5GaEIINiiY9708sFji@mail.gmail.com> 
	<AANLkTimDZx2T15GHnWck36bESESuK8n4u_wzzujyxvfE@mail.gmail.com> 
	<AANLkTilXbUSK5DbDb5RWboqEJLB6xAuPVwnDA21KsuZK@mail.gmail.com> 
	<B429D30727D34D35B0FA648556D337F3@MOBILEE1705>
	<AANLkTimGSAW5M9dmu2r70M2AM6jRMcSWZxjA__K7h8XW@mail.gmail.com> 
	<7350123A2AAC421587893A6BD6657C74@MOBILEE1705>
Message-ID: <AANLkTilXDGyz4MUWg3Mji1dKkeWCLKsRcd3JNPvZd97t@mail.gmail.com>

ok worked out the problem. Thanks for the help.  my stupid softphone does
not support dtmf receiving ti seems.  used normal phone to call and it all
worked.  grrrr

On Sat, May 15, 2010 at 11:39 AM, Madovsky <infos at madovsky.org> wrote:

>  I use it with fs_cli
>
> ----- Original Message -----
> *From:* Chris <freeswitch at gilligan.id.au>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Friday, May 14, 2010 9:33 PM
> *Subject:* Re: [Freeswitch-users] mod_managed or a script,sending DTMF and
> receiving DTMF
>
> do you happen to be using ti from inside mod_managed or via a different
> means?  Any chance of a small sample?  maybe i am doing something stupid.
>
> On Sat, May 15, 2010 at 11:22 AM, Madovsky <infos at madovsky.org> wrote:
>
>>  it works, I use it everyday
>>
>>   ----- Original Message -----
>> *From:* Chris <freeswitch at gilligan.id.au>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Sent:* Friday, May 14, 2010 8:52 PM
>> *Subject:* Re: [Freeswitch-users] mod_managed or a script,sending DTMF
>> and receiving DTMF
>>
>> sadly uuid_send_dtmf did not seem to work
>>
>>
>> On Fri, May 14, 2010 at 11:10 AM, Chris <freeswitch at gilligan.id.au>wrote:
>>
>>> Thanks so that is the only way?  I will have to do some tests and see if
>>> i can capture dtmf at the same time as i send when i do it that way.  Any
>>> other options?
>>>
>>>
>>> On Thu, May 13, 2010 at 10:50 PM, Phillip Jones <pjintheusa at gmail.com>wrote:
>>>
>>>> Try using http://wiki.freeswitch.org/wiki/Mod_commands#uuid_send_dtmf
>>>>
>>>>   On Thu, May 13, 2010 at 5:32 AM, Chris <freeswitch at gilligan.id.au>wrote:
>>>>
>>>>>  HI,
>>>>> I am currently working on a project in mod_managed and i am trying to
>>>>> discover the best way to meet my requirements.  i am hoping someone will
>>>>> have some ideas.  This could be implemented in one of the other scripting
>>>>> language if needed.
>>>>>
>>>>> What i am trying to do is reprogram a remote device via the phone.  The
>>>>> device takes commands in the form of DTMF tones and responds in different
>>>>> DTMF tones depending on success or failure. An example of the flow would be
>>>>>
>>>>> freeswitch -> 342523#
>>>>> device -> 1
>>>>> freeswitch -> 356789#
>>>>> device ->2
>>>>>
>>>>> device always responds with one digit and freeswitch sends many.
>>>>>
>>>>> I was hoping to do this with a database and mod_managed but i can't
>>>>> workout how to send the DTMF in mod_managed unless i user audio files for it
>>>>> which seems to be the wrong way to go to me. While i would prefer a
>>>>> mod_managed solution i will take anything i can find.
>>>>>
>>>>>
>>>>>
>>>>> Chris
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>  ------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Fri May 14 18:59:09 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 14 May 2010 21:59:09 -0400
Subject: [Freeswitch-users] mod_managed or a script,
	sending DTMF and 	receiving DTMF
References: <AANLkTikSTwOw1yiBGO9VrVDnsKtH7ySz8V2LxL1XUdru@mail.gmail.com>
	<AANLkTik1BYHsuHlfS9FAD3pEGS5GaEIINiiY9708sFji@mail.gmail.com>
	<AANLkTimDZx2T15GHnWck36bESESuK8n4u_wzzujyxvfE@mail.gmail.com>
	<AANLkTilXbUSK5DbDb5RWboqEJLB6xAuPVwnDA21KsuZK@mail.gmail.com>
	<B429D30727D34D35B0FA648556D337F3@MOBILEE1705><AANLkTimGSAW5M9dmu2r70M2AM6jRMcSWZxjA__K7h8XW@mail.gmail.com>
	<7350123A2AAC421587893A6BD6657C74@MOBILEE1705>
	<AANLkTilXDGyz4MUWg3Mji1dKkeWCLKsRcd3JNPvZd97t@mail.gmail.com>
Message-ID: <3D1F00DE01404A7E9D9C1942D401B982@MOBILEE1705>

read carefully RFC2833, RFC2976 and inband DTMF.
there are 3 major different ways to use DTMF. read the doc of your phone to know which DTMF it uses

F
  ----- Original Message ----- 
  From: Chris 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, May 14, 2010 9:45 PM
  Subject: Re: [Freeswitch-users] mod_managed or a script,sending DTMF and receiving DTMF


  ok worked out the problem. Thanks for the help.  my stupid softphone does not support dtmf receiving ti seems.  used normal phone to call and it all worked.  grrrr 


  On Sat, May 15, 2010 at 11:39 AM, Madovsky <infos at madovsky.org> wrote:

    I use it with fs_cli
      ----- Original Message ----- 
      From: Chris 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Friday, May 14, 2010 9:33 PM
      Subject: Re: [Freeswitch-users] mod_managed or a script,sending DTMF and receiving DTMF


      do you happen to be using ti from inside mod_managed or via a different means?  Any chance of a small sample?  maybe i am doing something stupid.


      On Sat, May 15, 2010 at 11:22 AM, Madovsky <infos at madovsky.org> wrote:

        it works, I use it everyday
          ----- Original Message ----- 
          From: Chris 
          To: freeswitch-users at lists.freeswitch.org 
          Sent: Friday, May 14, 2010 8:52 PM
          Subject: Re: [Freeswitch-users] mod_managed or a script,sending DTMF and receiving DTMF


          sadly uuid_send_dtmf did not seem to work 



          On Fri, May 14, 2010 at 11:10 AM, Chris <freeswitch at gilligan.id.au> wrote:

            Thanks so that is the only way?  I will have to do some tests and see if i can capture dtmf at the same time as i send when i do it that way.  Any other options? 



            On Thu, May 13, 2010 at 10:50 PM, Phillip Jones <pjintheusa at gmail.com> wrote:

              Try using http://wiki.freeswitch.org/wiki/Mod_commands#uuid_send_dtmf


              On Thu, May 13, 2010 at 5:32 AM, Chris <freeswitch at gilligan.id.au> wrote:

                HI, 
                I am currently working on a project in mod_managed and i am trying to discover the best way to meet my requirements.  i am hoping someone will have some ideas.  This could be implemented in one of the other scripting language if needed.


                What i am trying to do is reprogram a remote device via the phone.  The device takes commands in the form of DTMF tones and responds in different DTMF tones depending on success or failure. An example of the flow would be


                freeswitch -> 342523#
                device -> 1
                freeswitch -> 356789#
                device ->2


                device always responds with one digit and freeswitch sends many.


                I was hoping to do this with a database and mod_managed but i can't workout how to send the DTMF in mod_managed unless i user audio files for it which seems to be the wrong way to go to me. While i would prefer a mod_managed solution i will take anything i can find.






                Chris




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




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








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


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



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


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



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


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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From freeswitch at gilligan.id.au  Fri May 14 19:11:09 2010
From: freeswitch at gilligan.id.au (Chris)
Date: Sat, 15 May 2010 12:11:09 +1000
Subject: [Freeswitch-users] mod_managed or a script,
	sending DTMF and 	receiving DTMF
In-Reply-To: <3D1F00DE01404A7E9D9C1942D401B982@MOBILEE1705>
References: <AANLkTikSTwOw1yiBGO9VrVDnsKtH7ySz8V2LxL1XUdru@mail.gmail.com> 
	<AANLkTik1BYHsuHlfS9FAD3pEGS5GaEIINiiY9708sFji@mail.gmail.com> 
	<AANLkTimDZx2T15GHnWck36bESESuK8n4u_wzzujyxvfE@mail.gmail.com> 
	<AANLkTilXbUSK5DbDb5RWboqEJLB6xAuPVwnDA21KsuZK@mail.gmail.com> 
	<B429D30727D34D35B0FA648556D337F3@MOBILEE1705>
	<AANLkTimGSAW5M9dmu2r70M2AM6jRMcSWZxjA__K7h8XW@mail.gmail.com> 
	<7350123A2AAC421587893A6BD6657C74@MOBILEE1705>
	<AANLkTilXDGyz4MUWg3Mji1dKkeWCLKsRcd3JNPvZd97t@mail.gmail.com> 
	<3D1F00DE01404A7E9D9C1942D401B982@MOBILEE1705>
Message-ID: <AANLkTilIvDqbZ0P1PKDo7z_79H8bIUP6Y_7BkMLubKws@mail.gmail.com>

would you happen to know the default one in freeswitch? My system is for
landlines anyway and that seems to work.  the voip phone was just for free
testing so i did not have to pay for calls.

On Sat, May 15, 2010 at 11:59 AM, Madovsky <infos at madovsky.org> wrote:

>  read carefully RFC2833, RFC2976 and inband DTMF.
> there are 3 major different ways to use DTMF. read the doc of your phone to
> know which DTMF it uses
>
> F
>
> ----- Original Message -----
> *From:* Chris <freeswitch at gilligan.id.au>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Friday, May 14, 2010 9:45 PM
> *Subject:* Re: [Freeswitch-users] mod_managed or a script,sending DTMF and
> receiving DTMF
>
> ok worked out the problem. Thanks for the help.  my stupid softphone does
> not support dtmf receiving ti seems.  used normal phone to call and it all
> worked.  grrrr
>
> On Sat, May 15, 2010 at 11:39 AM, Madovsky <infos at madovsky.org> wrote:
>
>>  I use it with fs_cli
>>
>>   ----- Original Message -----
>> *From:* Chris <freeswitch at gilligan.id.au>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Sent:* Friday, May 14, 2010 9:33 PM
>> *Subject:* Re: [Freeswitch-users] mod_managed or a script,sending DTMF
>> and receiving DTMF
>>
>> do you happen to be using ti from inside mod_managed or via a different
>> means?  Any chance of a small sample?  maybe i am doing something stupid.
>>
>> On Sat, May 15, 2010 at 11:22 AM, Madovsky <infos at madovsky.org> wrote:
>>
>>>  it works, I use it everyday
>>>
>>>   ----- Original Message -----
>>> *From:* Chris <freeswitch at gilligan.id.au>
>>> *To:* freeswitch-users at lists.freeswitch.org
>>> *Sent:* Friday, May 14, 2010 8:52 PM
>>> *Subject:* Re: [Freeswitch-users] mod_managed or a script,sending DTMF
>>> and receiving DTMF
>>>
>>> sadly uuid_send_dtmf did not seem to work
>>>
>>>
>>> On Fri, May 14, 2010 at 11:10 AM, Chris <freeswitch at gilligan.id.au>wrote:
>>>
>>>> Thanks so that is the only way?  I will have to do some tests and see if
>>>> i can capture dtmf at the same time as i send when i do it that way.  Any
>>>> other options?
>>>>
>>>>
>>>> On Thu, May 13, 2010 at 10:50 PM, Phillip Jones <pjintheusa at gmail.com>wrote:
>>>>
>>>>> Try using http://wiki.freeswitch.org/wiki/Mod_commands#uuid_send_dtmf
>>>>>
>>>>>   On Thu, May 13, 2010 at 5:32 AM, Chris <freeswitch at gilligan.id.au>wrote:
>>>>>
>>>>>>  HI,
>>>>>> I am currently working on a project in mod_managed and i am trying to
>>>>>> discover the best way to meet my requirements.  i am hoping someone will
>>>>>> have some ideas.  This could be implemented in one of the other scripting
>>>>>> language if needed.
>>>>>>
>>>>>> What i am trying to do is reprogram a remote device via the phone.
>>>>>>  The device takes commands in the form of DTMF tones and responds in
>>>>>> different DTMF tones depending on success or failure. An example of the flow
>>>>>> would be
>>>>>>
>>>>>> freeswitch -> 342523#
>>>>>> device -> 1
>>>>>> freeswitch -> 356789#
>>>>>> device ->2
>>>>>>
>>>>>> device always responds with one digit and freeswitch sends many.
>>>>>>
>>>>>> I was hoping to do this with a database and mod_managed but i can't
>>>>>> workout how to send the DTMF in mod_managed unless i user audio files for it
>>>>>> which seems to be the wrong way to go to me. While i would prefer a
>>>>>> mod_managed solution i will take anything i can find.
>>>>>>
>>>>>>
>>>>>>
>>>>>> Chris
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>  ------------------------------
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>  ------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Fri May 14 19:18:32 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 14 May 2010 22:18:32 -0400
Subject: [Freeswitch-users] mod_managed or a script,
	sending DTMF and 	receiving DTMF
References: <AANLkTikSTwOw1yiBGO9VrVDnsKtH7ySz8V2LxL1XUdru@mail.gmail.com>
	<AANLkTik1BYHsuHlfS9FAD3pEGS5GaEIINiiY9708sFji@mail.gmail.com>
	<AANLkTimDZx2T15GHnWck36bESESuK8n4u_wzzujyxvfE@mail.gmail.com>
	<AANLkTilXbUSK5DbDb5RWboqEJLB6xAuPVwnDA21KsuZK@mail.gmail.com>
	<B429D30727D34D35B0FA648556D337F3@MOBILEE1705><AANLkTimGSAW5M9dmu2r70M2AM6jRMcSWZxjA__K7h8XW@mail.gmail.com>
	<7350123A2AAC421587893A6BD6657C74@MOBILEE1705><AANLkTilXDGyz4MUWg3Mji1dKkeWCLKsRcd3JNPvZd97t@mail.gmail.com>
	<3D1F00DE01404A7E9D9C1942D401B982@MOBILEE1705>
	<AANLkTilIvDqbZ0P1PKDo7z_79H8bIUP6Y_7BkMLubKws@mail.gmail.com>
Message-ID: <00054046A7FA4FE59A5C5AB41C43382C@MOBILEE1705>

you can set it into your sip_profiles 
  ----- Original Message ----- 
  From: Chris 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, May 14, 2010 10:11 PM
  Subject: Re: [Freeswitch-users] mod_managed or a script,sending DTMF and receiving DTMF


  would you happen to know the default one in freeswitch? My system is for landlines anyway and that seems to work.  the voip phone was just for free testing so i did not have to pay for calls.


  On Sat, May 15, 2010 at 11:59 AM, Madovsky <infos at madovsky.org> wrote:

    read carefully RFC2833, RFC2976 and inband DTMF.
    there are 3 major different ways to use DTMF. read the doc of your phone to know which DTMF it uses

    F
      ----- Original Message ----- 
      From: Chris 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Friday, May 14, 2010 9:45 PM
      Subject: Re: [Freeswitch-users] mod_managed or a script,sending DTMF and receiving DTMF


      ok worked out the problem. Thanks for the help.  my stupid softphone does not support dtmf receiving ti seems.  used normal phone to call and it all worked.  grrrr 


      On Sat, May 15, 2010 at 11:39 AM, Madovsky <infos at madovsky.org> wrote:

        I use it with fs_cli
          ----- Original Message ----- 
          From: Chris 
          To: freeswitch-users at lists.freeswitch.org 
          Sent: Friday, May 14, 2010 9:33 PM
          Subject: Re: [Freeswitch-users] mod_managed or a script,sending DTMF and receiving DTMF


          do you happen to be using ti from inside mod_managed or via a different means?  Any chance of a small sample?  maybe i am doing something stupid.


          On Sat, May 15, 2010 at 11:22 AM, Madovsky <infos at madovsky.org> wrote:

            it works, I use it everyday
              ----- Original Message ----- 
              From: Chris 
              To: freeswitch-users at lists.freeswitch.org 
              Sent: Friday, May 14, 2010 8:52 PM
              Subject: Re: [Freeswitch-users] mod_managed or a script,sending DTMF and receiving DTMF


              sadly uuid_send_dtmf did not seem to work 



              On Fri, May 14, 2010 at 11:10 AM, Chris <freeswitch at gilligan.id.au> wrote:

                Thanks so that is the only way?  I will have to do some tests and see if i can capture dtmf at the same time as i send when i do it that way.  Any other options? 



                On Thu, May 13, 2010 at 10:50 PM, Phillip Jones <pjintheusa at gmail.com> wrote:

                  Try using http://wiki.freeswitch.org/wiki/Mod_commands#uuid_send_dtmf


                  On Thu, May 13, 2010 at 5:32 AM, Chris <freeswitch at gilligan.id.au> wrote:

                    HI, 
                    I am currently working on a project in mod_managed and i am trying to discover the best way to meet my requirements.  i am hoping someone will have some ideas.  This could be implemented in one of the other scripting language if needed.


                    What i am trying to do is reprogram a remote device via the phone.  The device takes commands in the form of DTMF tones and responds in different DTMF tones depending on success or failure. An example of the flow would be


                    freeswitch -> 342523#
                    device -> 1
                    freeswitch -> 356789#
                    device ->2


                    device always responds with one digit and freeswitch sends many.


                    I was hoping to do this with a database and mod_managed but i can't workout how to send the DTMF in mod_managed unless i user audio files for it which seems to be the wrong way to go to me. While i would prefer a mod_managed solution i will take anything i can find.






                    Chris




                    _______________________________________________
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                    FreeSWITCH-users at lists.freeswitch.org
                    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
                    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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                  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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------------------------------------------------------------------


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






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


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  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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-------------- next part --------------
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From christian.loeschenkohl at xpirio.com  Sat May 15 00:01:30 2010
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Sat, 15 May 2010 09:01:30 +0200
Subject: [Freeswitch-users] blind transfer and outbound socket
In-Reply-To: <AANLkTikpQXPohk12_NENaDJCFrQZy7QET9mTQnLHVNIM@mail.gmail.com>
References: <4BE87E88.2000106@xpirio.com>	<AANLkTikcfzQS6uuSHiCAdllO5ApvAiUT_l1aXMSU3HLT@mail.gmail.com>	<4BE88515.30105@xpirio.com>	<AANLkTinftpVgycyYVPLh6-xS9aLPFnReMz3AjAVtIfDh@mail.gmail.com>	<4BE88C98.5050107@xpirio.com>	<3A758DA1-0AD2-401E-9FDF-BD7A4258DC03@gmail.com>
	<AANLkTikpQXPohk12_NENaDJCFrQZy7QET9mTQnLHVNIM@mail.gmail.com>
Message-ID: <4BEE46CA.2080101@xpirio.com>

hi

thank very much for this hint, but i run freeswitch as root

ps -ef shows
root      9841     1 13 May13 ?        05:12:11 /opt/freeswitch/bin/freeswitch -nc -nonat

i couldn't help myself otherwise so i use mod_perl for this task, all information
needed was present in variables so i could do the transfer with a few lines.

br


Rupa Schomaker wrote:

> The cause is a constrained stack space which can't be made larger if you 
> aren't running FS as super user (you can reduce your stack limit, but 
> once reduced can't increase it).  
> 
> I've also run into this in other situations.  I record calls as .wav 
> files but then convert to mp3.  Initially, I tried to do the conversion 
> via a system() api in the hangup hook.  But lame would die due to 
> insufficient stack space.  Instead I now just move the .wav file to a 
> processing directory where a cron job picks it up, extracts the metadata 
> from the wav and then converts to mp3 applying the metadata.  
> 
> You could look at using an alternative sendmail.  Some smtp servers have 
> a sendmail binary that is very lightweight and would not suffer in a low 
> stack space environment.  postfix for instance just drops the email into 
> the drop directory (which is then picked up by the postfix daemon).  
> 
> The real sendmail on the other hand is going to do everything 
> in-process.  So it needs to do a lot more work to do that email delivery.
> 
> On Tue, May 11, 2010 at 12:56 AM, David Ponzone <david.ponzone at gmail.com 
> <mailto:david.ponzone at gmail.com>> wrote:
> 
>     Christian,
> 
>     if you send the mail using a shell call to "sendmail", it's possible
>     it won't work.
>     Some weeks ago, I and another person have noticed that when you do
>     that, sendmail crashes.
>     I don't know what was the cause and if it is fixed.
> 
>     Writing to console would be a simpler way to know your script is
>     executed.
> 
>     David Ponzone  Direction Technique
>     email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>     tel:      01 74 03 18 97
>     gsm:   06 66 98 76 34
> 
>     Service Client IPeva
>     tel:      0811 46 26 26
>     www.ipeva.fr  -   www.ipeva-studio.com
> 
>     /Ce message et toutes les pi?ces jointes sont confidentiels et
>     ?tablis ? l'intention exclusive de ses destinataires. Toute
>     utilisation ou diffusion non autoris?e est interdite. Tout message
>     ?lectronique est susceptible d'alt?ration. /*/IPeva/*/ d?cline toute
>     responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou
>     falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le
>     d?truire imm?diatement et d'avertir l'exp?diteur./
>     /
>     /
> 
> 
> 
>     Le 11/05/2010 ? 00:45, Christian L?schenkohl a ?crit :
> 
>>     yes, and this is my problem
>>     it looks normal but the scripts are not executed (i send a email
>>     in the fist line of
>>     the script and there is no mail send to me here).
>>     the call is hung up without much comment, the dialplan runs
>>     through until the hangup -
>>     without the bridge command executed from the script.
>>
>>     attended transfer is here, with this it works
>>     http://pastebin.freeswitch.org/12959
>>
>>     the scripts do also work if i call the extension directly.
>>     the extension 50 also rings if i put the bridge in the dialplan -
>>     my only problem here
>>     is that it looks like the script is called but it is never executed.
>>
>>     br
>>
>>
>>
>>     Anthony Minessale wrote:
>>
>>>     it looks like it works fine to me.
>>>
>>>     the call transfers back to the dialplan where it again executes your
>>>     socket app and that app either does nothing
>>>     or hangs up.
>>>
>>>     See line 789 of your pastebin.
>>>
>>>
>>>     2010/5/10 Christian L?schenkohl
>>>     <christian.loeschenkohl at xpirio.com
>>>     <mailto:christian.loeschenkohl at xpirio.com>
>>>     <mailto:christian.loeschenkohl at xpirio.com
>>>     <mailto:christian.loeschenkohl at xpirio.com>>>
>>>
>>>        the scripts do some database lookups (setting callers name) and do
>>>        find the called
>>>        user e.g. expand the called 50 to 50 at customer.domain.com
>>>     <mailto:50 at customer.domain.com>
>>>        <mailto:50 at customer.domain.com
>>>     <mailto:50 at customer.domain.com>> and finaly do the call bridge.
>>>
>>>        the transfer itself is done by the phone/softclient with a refer
>>>        packet, as far as i can see.
>>>
>>>        a debug trace is here
>>>        http://pastebin.freeswitch.org/12958
>>>
>>>        br
>>>
>>>        Anthony Minessale wrote:
>>>
>>>>     what exactly does your script call to try to make it transfer?
>>>>
>>>>     2010/5/10 Christian L?schenkohl
>>>        <christian.loeschenkohl at xpirio.com
>>>     <mailto:christian.loeschenkohl at xpirio.com>
>>>        <mailto:christian.loeschenkohl at xpirio.com
>>>     <mailto:christian.loeschenkohl at xpirio.com>>
>>>>     <mailto:christian.loeschenkohl at xpirio.com
>>>>     <mailto:christian.loeschenkohl at xpirio.com>
>>>        <mailto:christian.loeschenkohl at xpirio.com
>>>     <mailto:christian.loeschenkohl at xpirio.com>>>>
>>>>
>>>>        hello
>>>>
>>>>        i have a problem with a freeswitch hosted pbx system.
>>>>        when i call in to a extension (e.g. 40) an answer the call
>>>        and then
>>>>        do a blind transfer to extension 50 it
>>>>        doesn't work.
>>>>        the outbound socket script do not work or do not get called
>>>        in this
>>>>        case (no mail from the script).
>>>>
>>>>        on the other hand if i do a attended transfer it works as it
>>>        should
>>>>        (sending mail in the script works).
>>>>        the same context, the same scripts.
>>>>
>>>>        we use snom 320 phones, the sip refer looks normal.
>>>>        blind transfer also doesn't work with a softclient (sjphone)
>>>>
>>>>        where could i start?
>>>>
>>>>        br
>>>>
>>>>        --
>>>>        Ing. Christian L?schenkohl
>>>>        Technische Leitung, Forschung & Entwicklung VoIP
>>>>
>>>>        xpirio
>>>>        Telekommunikation & Service GmbH
>>>>        Lakeside B04
>>>>        9020 Klagenfurt
>>>>        Austria
>>>>
>>>>        T  +43 (0) 5 77 11 - 1000
>>>>        F  +43 (0) 5 77 11 - 1002
>>>>        E  christian.loeschenkohl at xpirio.com
>>>>     <mailto:christian.loeschenkohl at xpirio.com>
>>>        <mailto:christian.loeschenkohl at xpirio.com
>>>     <mailto:christian.loeschenkohl at xpirio.com>>
>>>>        <mailto:christian.loeschenkohl at xpirio.com
>>>>     <mailto:christian.loeschenkohl at xpirio.com>
>>>        <mailto:christian.loeschenkohl at xpirio.com
>>>     <mailto:christian.loeschenkohl at xpirio.com>>>
>>>>
>>>>        _______________________________________________
>>>>        FreeSWITCH-users mailing list
>>>>        FreeSWITCH-users at lists.freeswitch.org
>>>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>        <mailto:FreeSWITCH-users at lists.freeswitch.org
>>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>>>>        <mailto:FreeSWITCH-users at lists.freeswitch.org
>>>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>        <mailto:FreeSWITCH-users at lists.freeswitch.org
>>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>>
>>>>        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>
>>>        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>        http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>>
>>>>     --
>>>>     Anthony Minessale II
>>>>
>>>>     FreeSWITCH http://www.freeswitch.org/
>>>>     ClueCon http://www.cluecon.com/
>>>>     Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>>     AIM: anthm
>>>>     MSN:anthony_minessale at hotmail.com
>>>>     <mailto:MSN%3Aanthony_minessale at hotmail.com>
>>>        <mailto:MSN%3Aanthony_minessale at hotmail.com
>>>     <mailto:MSN%253Aanthony_minessale at hotmail.com>>
>>>>     <mailto:MSN%3Aanthony_minessale at hotmail.com
>>>>     <mailto:MSN%253Aanthony_minessale at hotmail.com>
>>>        <mailto:MSN%253Aanthony_minessale at hotmail.com
>>>     <mailto:MSN%25253Aanthony_minessale at hotmail.com>>>
>>>>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>>     <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>>>        <mailto:PAYPAL%3Aanthony.minessale at gmail.com
>>>     <mailto:PAYPAL%253Aanthony.minessale at gmail.com>>
>>>>     <mailto:PAYPAL%3Aanthony.minessale at gmail.com
>>>>     <mailto:PAYPAL%253Aanthony.minessale at gmail.com>
>>>        <mailto:PAYPAL%253Aanthony.minessale at gmail.com
>>>     <mailto:PAYPAL%25253Aanthony.minessale at gmail.com>>>
>>>>     IRC: irc.freenode.net <http://irc.freenode.net>
>>>>     <http://irc.freenode.net>
>>>        <http://irc.freenode.net> #freeswitch
>>>>
>>>>     FreeSWITCH Developer Conference
>>>>     sip:888 at conference.freeswitch.org
>>>>     <mailto:sip%3A888 at conference.freeswitch.org>
>>>        <mailto:sip%3A888 at conference.freeswitch.org
>>>     <mailto:sip%253A888 at conference.freeswitch.org>>
>>>>     <mailto:sip%3A888 at conference.freeswitch.org
>>>>     <mailto:sip%253A888 at conference.freeswitch.org>
>>>        <mailto:sip%253A888 at conference.freeswitch.org
>>>     <mailto:sip%25253A888 at conference.freeswitch.org>>>
>>>>     googletalk:conf+888 at conference.freeswitch.org
>>>>     <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>        <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org
>>>     <mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org>>
>>>>     <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org
>>>>     <mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org>
>>>        <mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org
>>>     <mailto:googletalk%25253Aconf%25252B888 at conference.freeswitch.org>>>
>>>>     pstn:+19193869900
>>>>
>>>>
>>>>
>>>        ------------------------------------------------------------------------
>>>>
>>>>     _______________________________________________
>>>>     FreeSWITCH-users mailing list
>>>>     FreeSWITCH-users at lists.freeswitch.org
>>>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>        <mailto:FreeSWITCH-users at lists.freeswitch.org
>>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>
>>>        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>     http://www.freeswitch.org
>>>
>>>
>>>        --
>>>        Ing. Christian L?schenkohl
>>>        Technische Leitung, Forschung & Entwicklung VoIP
>>>
>>>        xpirio
>>>        Telekommunikation & Service GmbH
>>>        Lakeside B04
>>>        9020 Klagenfurt
>>>        Austria
>>>
>>>        T  +43 (0) 5 77 11 - 1000
>>>        F  +43 (0) 5 77 11 - 1002
>>>        E  christian.loeschenkohl at xpirio.com
>>>     <mailto:christian.loeschenkohl at xpirio.com>
>>>        <mailto:christian.loeschenkohl at xpirio.com
>>>     <mailto:christian.loeschenkohl at xpirio.com>>
>>>
>>>        _______________________________________________
>>>        FreeSWITCH-users mailing list
>>>        FreeSWITCH-users at lists.freeswitch.org
>>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>        <mailto:FreeSWITCH-users at lists.freeswitch.org
>>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>>>        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>        UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>        http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>>     -- 
>>>     Anthony Minessale II
>>>
>>>     FreeSWITCH http://www.freeswitch.org/
>>>     ClueCon http://www.cluecon.com/
>>>     Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>>     AIM: anthm
>>>     MSN:anthony_minessale at hotmail.com
>>>     <mailto:MSN%3Aanthony_minessale at hotmail.com>
>>>     <mailto:MSN%3Aanthony_minessale at hotmail.com
>>>     <mailto:MSN%253Aanthony_minessale at hotmail.com>>
>>>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>>     <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>>>     <mailto:PAYPAL%3Aanthony.minessale at gmail.com
>>>     <mailto:PAYPAL%253Aanthony.minessale at gmail.com>>
>>>     IRC: irc.freenode.net <http://irc.freenode.net>
>>>     <http://irc.freenode.net> #freeswitch
>>>
>>>     FreeSWITCH Developer Conference
>>>     sip:888 at conference.freeswitch.org
>>>     <mailto:sip%3A888 at conference.freeswitch.org>
>>>     <mailto:sip%3A888 at conference.freeswitch.org
>>>     <mailto:sip%253A888 at conference.freeswitch.org>>
>>>     googletalk:conf+888 at conference.freeswitch.org
>>>     <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>     <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org
>>>     <mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org>>
>>>     pstn:+19193869900
>>>
>>>
>>>     ------------------------------------------------------------------------
>>>
>>>     _______________________________________________
>>>     FreeSWITCH-users mailing list
>>>     FreeSWITCH-users at lists.freeswitch.org
>>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     http://www.freeswitch.org
>>
>>
>>     -- 
>>     Ing. Christian L?schenkohl
>>     Technische Leitung, Forschung & Entwicklung VoIP
>>
>>     xpirio
>>     Telekommunikation & Service GmbH
>>     Lakeside B04
>>     9020 Klagenfurt
>>     Austria
>>
>>     T  +43 (0) 5 77 11 - 1000
>>     F  +43 (0) 5 77 11 - 1002
>>     E  christian.loeschenkohl at xpirio.com
>>     <mailto:christian.loeschenkohl at xpirio.com>
>>
>>     _______________________________________________
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
> 
> 
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> -Rupa
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com


From jmesquita at freeswitch.org  Sat May 15 00:30:57 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sat, 15 May 2010 04:30:57 -0300
Subject: [Freeswitch-users] openvox PRI card for freeswitch
In-Reply-To: <E867C025-C339-49EA-8D69-25EB3DB58233@gmail.com>
References: <AANLkTilvzyhuyQO09FyNv8QIwQNAgDlUpPA6YbukbsUH@mail.gmail.com>
	<321755B2-85FD-4BB6-B6D8-25BFBC0C5AA6@jerris.com>
	<AANLkTilaF8g9io8WICUQrO7cWRmU4FPScDlIX3uyIFiC@mail.gmail.com>
	<69A6B994-FCE3-497D-A43E-2B4F83419754@gmail.com>
	<AANLkTikxdU67WA3MWO1TZR_8FHxZIxZth9h0u-hg_9Ap@mail.gmail.com>
	<E867C025-C339-49EA-8D69-25EB3DB58233@gmail.com>
Message-ID: <AANLkTikDyng8FqcMLfnEfYxNNmBjmnUHvprhRj3KTowq@mail.gmail.com>

David, I couldn't help but notice that you are in France. Are we talking
about T1 or E1? If you are talking about E1 and want a solution you should
get in touch with me to check on the Khomp boards. We *might* be able to
send you a lab board so you can try it out. They are NOT as cheap as OpenVox
as this is not the goal but they loose for absolutely nothing on the Sangoma
cards. Quality do have its costs, ya know?

Regards,

Jo?o Mesquita


On Fri, May 14, 2010 at 9:59 AM, David Ponzone <david.ponzone at gmail.com>wrote:

> 4 ports is not an option as my goal is to find a cheaper alternative to the
> Sangoma A101D/DE so I can build a reliable but cheap 1PRI/SIP gateway.
>
> It seems in the end, Sangoma has no alternative.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 14/05/2010 ? 14:05, Krzysztof Drewicz a ?crit :
>
> 2010/5/14 David Ponzone <david.ponzone at gmail.com>:
>
> Krzysztof,
>
> May I use the opportunity to ask your opinion on those cards ?
>
> They are strangely inexpensive (particularly the D110/115), and I can't
> find
>
> any mention of HW EC on their website.
>
>
> The one-port are not very good (no hw ec, even as an option). But for
> simple use as faxserver without any 'big' traffic they seem to work
> fine.
> But still, one card per server :(
>
> Can you confirm this ?
>
> Any issue using it with FreeSWITCH ?
>
>
> Hm, whould recomend the 4 port, they got hw ec (as an option but buy
> hw ec with card, even if you plan to use it afterwards, some times
> there is card with older firmware and hw ec has new firmware and they
> don't match).
> Card are recognized as Digium cards, they work :) just don't put too
> much traffic on them (one 4 port card per one good box seems to be
> OK).
>
> I buy them from some local internet-based shop and never had any
> problem (like card not working or not compatible with server mainboard
> etc...).
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at gmail.com  Sat May 15 00:51:33 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Sat, 15 May 2010 09:51:33 +0200
Subject: [Freeswitch-users] openvox PRI card for freeswitch
In-Reply-To: <AANLkTikDyng8FqcMLfnEfYxNNmBjmnUHvprhRj3KTowq@mail.gmail.com>
References: <AANLkTilvzyhuyQO09FyNv8QIwQNAgDlUpPA6YbukbsUH@mail.gmail.com>
	<321755B2-85FD-4BB6-B6D8-25BFBC0C5AA6@jerris.com>
	<AANLkTilaF8g9io8WICUQrO7cWRmU4FPScDlIX3uyIFiC@mail.gmail.com>
	<69A6B994-FCE3-497D-A43E-2B4F83419754@gmail.com>
	<AANLkTikxdU67WA3MWO1TZR_8FHxZIxZth9h0u-hg_9Ap@mail.gmail.com>
	<E867C025-C339-49EA-8D69-25EB3DB58233@gmail.com>
	<AANLkTikDyng8FqcMLfnEfYxNNmBjmnUHvprhRj3KTowq@mail.gmail.com>
Message-ID: <986A6E9F-3BA4-46B8-848A-10415D193128@gmail.com>

Jo?o,

Thank you.
I checked out the Khomp K1E1-SPX card.
If I am not mistaken, the public price is around 1400? which is twice  
the price of Sangoma.
Am I missing something ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 15/05/2010 ? 09:30, Jo?o Mesquita a ?crit :

> David, I couldn't help but notice that you are in France. Are we  
> talking about T1 or E1? If you are talking about E1 and want a  
> solution you should get in touch with me to check on the Khomp  
> boards. We might be able to send you a lab board so you can try it  
> out. They are NOT as cheap as OpenVox as this is not the goal but  
> they loose for absolutely nothing on the Sangoma cards. Quality do  
> have its costs, ya know?
>
> Regards,
>
> Jo?o Mesquita
>
>
> On Fri, May 14, 2010 at 9:59 AM, David Ponzone <david.ponzone at gmail.com 
> > wrote:
> 4 ports is not an option as my goal is to find a cheaper alternative  
> to the Sangoma A101D/DE so I can build a reliable but cheap 1PRI/SIP  
> gateway.
>
> It seems in the end, Sangoma has no alternative.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et  
> ?tablis ? l'intention exclusive de ses destinataires. Toute  
> utilisation ou diffusion non autoris?e est interdite. Tout message  
> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou  
> falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le  
> d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
>
> Le 14/05/2010 ? 14:05, Krzysztof Drewicz a ?crit :
>
>> 2010/5/14 David Ponzone <david.ponzone at gmail.com>:
>>> Krzysztof,
>>> May I use the opportunity to ask your opinion on those cards ?
>>> They are strangely inexpensive (particularly the D110/115), and I  
>>> can't find
>>> any mention of HW EC on their website.
>>
>> The one-port are not very good (no hw ec, even as an option). But for
>> simple use as faxserver without any 'big' traffic they seem to work
>> fine.
>> But still, one card per server :(
>>
>>> Can you confirm this ?
>>> Any issue using it with FreeSWITCH ?
>>
>> Hm, whould recomend the 4 port, they got hw ec (as an option but buy
>> hw ec with card, even if you plan to use it afterwards, some times
>> there is card with older firmware and hw ec has new firmware and they
>> don't match).
>> Card are recognized as Digium cards, they work :) just don't put too
>> much traffic on them (one 4 port card per one good box seems to be
>> OK).
>>
>> I buy them from some local internet-based shop and never had any
>> problem (like card not working or not compatible with server  
>> mainboard
>> etc...).
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jmesquita at freeswitch.org  Sat May 15 01:02:29 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sat, 15 May 2010 05:02:29 -0300
Subject: [Freeswitch-users] openvox PRI card for freeswitch
In-Reply-To: <986A6E9F-3BA4-46B8-848A-10415D193128@gmail.com>
References: <AANLkTilvzyhuyQO09FyNv8QIwQNAgDlUpPA6YbukbsUH@mail.gmail.com>
	<321755B2-85FD-4BB6-B6D8-25BFBC0C5AA6@jerris.com>
	<AANLkTilaF8g9io8WICUQrO7cWRmU4FPScDlIX3uyIFiC@mail.gmail.com>
	<69A6B994-FCE3-497D-A43E-2B4F83419754@gmail.com>
	<AANLkTikxdU67WA3MWO1TZR_8FHxZIxZth9h0u-hg_9Ap@mail.gmail.com>
	<E867C025-C339-49EA-8D69-25EB3DB58233@gmail.com>
	<AANLkTikDyng8FqcMLfnEfYxNNmBjmnUHvprhRj3KTowq@mail.gmail.com>
	<986A6E9F-3BA4-46B8-848A-10415D193128@gmail.com>
Message-ID: <AANLkTikPywG4A008lxvvg1DDpyuQJAbRQjNaNxPtCZF0@mail.gmail.com>

Definitely! How/where did you check those prices? They are not even open to
the public as the company rather work with system integrators rather then
end users like OpenVox or Digium.

Maybe you are looking at prices in Reais (brazilian currency)?

Contact me offlist of you are interested on the boards and we can have a
chat over this next week, maybe?

Regards,
Jo?o Mesquita



On Sat, May 15, 2010 at 4:51 AM, David Ponzone <david.ponzone at gmail.com>wrote:

> Jo?o,
>
> Thank you.
> I checked out the Khomp K1E1-SPX card.
> If I am not mistaken, the public price is around 1400? which is twice the
> price of Sangoma.
> Am I missing something ?
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 15/05/2010 ? 09:30, Jo?o Mesquita a ?crit :
>
> David, I couldn't help but notice that you are in France. Are we talking
> about T1 or E1? If you are talking about E1 and want a solution you should
> get in touch with me to check on the Khomp boards. We *might* be able to
> send you a lab board so you can try it out. They are NOT as cheap as OpenVox
> as this is not the goal but they loose for absolutely nothing on the Sangoma
> cards. Quality do have its costs, ya know?
>
> Regards,
>
> Jo?o Mesquita
>
>
> On Fri, May 14, 2010 at 9:59 AM, David Ponzone <david.ponzone at gmail.com>wrote:
>
>> 4 ports is not an option as my goal is to find a cheaper alternative to
>> the Sangoma A101D/DE so I can build a reliable but cheap 1PRI/SIP gateway.
>>
>> It seems in the end, Sangoma has no alternative.
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 14/05/2010 ? 14:05, Krzysztof Drewicz a ?crit :
>>
>> 2010/5/14 David Ponzone <david.ponzone at gmail.com>:
>>
>> Krzysztof,
>>
>> May I use the opportunity to ask your opinion on those cards ?
>>
>> They are strangely inexpensive (particularly the D110/115), and I can't
>> find
>>
>> any mention of HW EC on their website.
>>
>>
>> The one-port are not very good (no hw ec, even as an option). But for
>> simple use as faxserver without any 'big' traffic they seem to work
>> fine.
>> But still, one card per server :(
>>
>> Can you confirm this ?
>>
>> Any issue using it with FreeSWITCH ?
>>
>>
>> Hm, whould recomend the 4 port, they got hw ec (as an option but buy
>> hw ec with card, even if you plan to use it afterwards, some times
>> there is card with older firmware and hw ec has new firmware and they
>> don't match).
>> Card are recognized as Digium cards, they work :) just don't put too
>> much traffic on them (one 4 port card per one good box seems to be
>> OK).
>>
>> I buy them from some local internet-based shop and never had any
>> problem (like card not working or not compatible with server mainboard
>> etc...).
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From babak.freeswitch at gmail.com  Sat May 15 03:14:03 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Sat, 15 May 2010 14:44:03 +0430
Subject: [Freeswitch-users] dll sym error
Message-ID: <AANLkTikfEtaFuN9xiVFDEa11yEnpsM7PgdTQ7h5xFqN-@mail.gmail.com>

Hi
I'm changing one of say modules, but when I try to load, it gives "dll sym
error"?
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From sean at obscuradigital.com  Sat May 15 09:03:04 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Sat, 15 May 2010 09:03:04 -0700
Subject: [Freeswitch-users] Mod directory question
Message-ID: <C81413C8.9FD1%sean@obscuradigital.com>

Hey list,

I had this working before but after upgrade it doesn?t.

Basically I setup the find by name feature for my ivr menu and it works
great, but now after entering the name the voice wav file spells out the
name.  Before it would use the recorded user name that the user created on
their phone.  Is there a setting I need in my dailplan to make this work
again.  I see the user created recordings on the FS server.

Thanks
Sean
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From larclap at yahoo.com  Sat May 15 09:52:58 2010
From: larclap at yahoo.com (Lars Zeb)
Date: Sat, 15 May 2010 09:52:58 -0700
Subject: [Freeswitch-users] Digium TDM400P
Message-ID: <005401caf44f$12a509c0$37ef1d40$@yahoo.com>

I had a Digium TDM400P which I used successfully in another box running
SwitchVox. I put it in my FreeSWITCH box.

I haven't started installing the zaptel software because 'lspci' does not
list the Digium card. Should it? I don't want to install the software if the
board's dead.

If it is dead, should I try to get it repaired or perhaps buy a Patton
M-ATA? I have a Viking E-40 Entry Phone which I would like to use with
FreeSWITCH.

The Viking along with the Digium board did work on the SwitchVox.

Thanks, Lars



From jcasale at activenetwerx.com  Sat May 15 11:04:20 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Sat, 15 May 2010 18:04:20 +0000
Subject: [Freeswitch-users] Digium TDM400P
In-Reply-To: <005401caf44f$12a509c0$37ef1d40$@yahoo.com>
References: <005401caf44f$12a509c0$37ef1d40$@yahoo.com>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E9A0A830@prato.activenetwerx.local>

>I haven't started installing the zaptel software because 'lspci' does not
>list the Digium card. Should it? I don't want to install the software if the
>board's dead.

It sure does:

# lspci |grep Digium
04:00.0 Ethernet controller: Digium, Inc. TDM400P (rev 11)

>If it is dead, should I try to get it repaired or perhaps buy a Patton
>M-ATA? I have a Viking E-40 Entry Phone which I would like to use with
>FreeSWITCH.

That card works _perfect_ in Asterisk obviously but I have yet to make it work
as well in FS.

>The Viking along with the Digium board did work on the SwitchVox.

Move the position of the board amongst your available peripheral slots, there
are some known compat issues, check Digiums KB.

jlc


From toqeer83 at gmail.com  Sat May 15 11:12:11 2010
From: toqeer83 at gmail.com (toqeer ali)
Date: Sat, 15 May 2010 11:12:11 -0700
Subject: [Freeswitch-users] Hunting
In-Reply-To: <AANLkTimJhhBIxFOuMcWjaa9SHnwAvQ7yvUzOl5e_YrJC@mail.gmail.com>
References: <AANLkTinW51XkJQAqDdI-D8760rrVOnMfaFomGjzLH7-L@mail.gmail.com>
	<4BED7112.1040701@communicatefreely.net>
	<AANLkTikEr7gTTHWvYVnNTPhu7R6FzCohsg14pMSUE_UR@mail.gmail.com>
	<AANLkTimJhhBIxFOuMcWjaa9SHnwAvQ7yvUzOl5e_YrJC@mail.gmail.com>
Message-ID: <AANLkTilcNAYIjlv_AmGnrLURIM2kHtd78NSF4XNSQwZo@mail.gmail.com>

Thank you very much... it works fine

Thanks alot

On Fri, May 14, 2010 at 1:46 PM, Steven Ward <steve.d.ward at gmail.com> wrote:

> As for timing out a bridge attempt...
>
> Before you do the bridge, one option is to set the call_timeout channel
> variable to 60.
>
> But, if the destination endpoint sends early media, you'll also have to set
> ignore_early_media to true before you do the bridge, since call_timeout only
> applies until the call attempt is successful (it's successful when you
> either get early media or a true answer).
>
> You can check out
> http://wiki.freeswitch.org/wiki/Channel_Variables#Timeout_Related
>
> Of course, it's all about the details of what you're doing and why, but
> those options should provide a really good place to start.
>
>
>
> On Fri, May 14, 2010 at 12:39 PM, toqeer ali <toqeer83 at gmail.com> wrote:
>
>> Hi Tim St. Pierr
>>
>> Thanks you very much for your reply i got the point.
>>
>> But one question more please, can i also define the time for example for
>> 60 second one extension could not answer go to other one?
>>
>> Thanks
>>
>> On Fri, May 14, 2010 at 8:49 AM, Tim St. Pierre <
>> fs-list at communicatefreely.net> wrote:
>>
>>> Nope, no special module required.
>>>
>>> Before you call the bridge application, set continue_on_fail=true
>>> This will tell freeswitch to keep going in the dialplan if the first
>>> destination doesn't answer.
>>> After the first bridge application, add other entries that take try the
>>> next destination.  You could
>>> use the transfer app to try another extension as you described.
>>>
>>> -Tim
>>>
>>> toqeer ali wrote:
>>> > Hi all,
>>> >
>>> > I am new to freeswitch and want to implement Hunting.
>>> >
>>> > For example if call come to my freeswitch box where DID is configured
>>> to
>>> > an extension for example 1000 so if 1000 did not pick the phone then it
>>> > should be transfered to other extension or landline.
>>> >
>>> > Please give me a clue... Is there is any special module for that?
>>> >
>>> > Any help will highly appreciated.
>>> >
>>> > Thanks
>>> >
>>> > --
>>> > Toqeer Ali Syed
>>> >
>>> > Red Hat Certified Engineer
>>> > mob:     +92 321 9059916
>>> >
>>> >
>>> >
>>> ------------------------------------------------------------------------
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Toqeer Ali Syed
>>
>> Red Hat Certified Engineer
>> mob:     +92 321 9059916
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Toqeer Ali Syed

Red Hat Certified Engineer
mob:     +92 321 9059916
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From neilp at cs.stanford.edu  Sat May 15 12:52:50 2010
From: neilp at cs.stanford.edu (Neil Patel)
Date: Sat, 15 May 2010 12:52:50 -0700
Subject: [Freeswitch-users] span not defined error
Message-ID: <AANLkTik9DQcT5fpgcfb6KqFUZT4vXUuAjqTE5pFLxW-R@mail.gmail.com>

Hi All,

I am trying to dial out over my PRI, and am getting this error:

2010-05-16 01:01:21.452392 [CRIT] zap_io.c:1139 SPAN NOT DEFINED!
2010-05-16 01:01:21.452392 [ERR] mod_openzap.c:1154 No channels available
2010-05-16 01:01:21.452392 [ERR] switch_ivr_originate.c:2249 Cannot create
outgoing channel of type [OpenZAP] cause: [NORMAL_CIRCUIT_CONGESTION]


This is my openzap.conf:

[span wanpipe smg_prid]
name => smg_prid
trunk_type =>e1
b-channel => 1:1-15
b-channel => 1:17-31
trunk_type =>e1
b-channel => 2:1-15
b-channel => 2:17-31


This is my openzap.conf.xml:

<configuration name="openzap.conf" description="OpenZAP Configuration">
  <settings>
    <param name="debug" value="0"/>

  </settings>
  <boost_spans>
        <span id="smg_prid">
        <param name="dialplan" value="XML"/>
        <param name="context" value="default"/>
        </span>
  </boost_spans>

  <analog_spans>
        <span id="smg_prid">
        <param name="dialplan" value="XML"/>
        <param name="context" value="default"/>

        </span>
  </analog_spans>
</configuration>


And here is the lua code I'm using to dial out:

sessiondata = "OpenZAP/smg_prid/<num>"
new_session = freeswitch.Session(sessiondata)


What am I missing here?
Thanks,
Neil
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From moises.silva at gmail.com  Sat May 15 15:18:21 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Sat, 15 May 2010 18:18:21 -0400
Subject: [Freeswitch-users] span not defined error
In-Reply-To: <AANLkTik9DQcT5fpgcfb6KqFUZT4vXUuAjqTE5pFLxW-R@mail.gmail.com>
References: <AANLkTik9DQcT5fpgcfb6KqFUZT4vXUuAjqTE5pFLxW-R@mail.gmail.com>
Message-ID: <AANLkTin9wP0f74qEyA5GjtpVo7YacbS2C6VE9VnO31wj@mail.gmail.com>

Why do you have 2 spans in openzap.conf.xml with the same name, in both the
boost and analog sections?
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com


On Sat, May 15, 2010 at 3:52 PM, Neil Patel <neilp at cs.stanford.edu> wrote:

> Hi All,
>
> I am trying to dial out over my PRI, and am getting this error:
>
> 2010-05-16 01:01:21.452392 [CRIT] zap_io.c:1139 SPAN NOT DEFINED!
> 2010-05-16 01:01:21.452392 [ERR] mod_openzap.c:1154 No channels available
> 2010-05-16 01:01:21.452392 [ERR] switch_ivr_originate.c:2249 Cannot create
> outgoing channel of type [OpenZAP] cause: [NORMAL_CIRCUIT_CONGESTION]
>
>
> This is my openzap.conf:
>
> [span wanpipe smg_prid]
> name => smg_prid
> trunk_type =>e1
> b-channel => 1:1-15
> b-channel => 1:17-31
> trunk_type =>e1
> b-channel => 2:1-15
> b-channel => 2:17-31
>
>
> This is my openzap.conf.xml:
>
> <configuration name="openzap.conf" description="OpenZAP Configuration">
>   <settings>
>     <param name="debug" value="0"/>
>
>   </settings>
>   <boost_spans>
>         <span id="smg_prid">
>         <param name="dialplan" value="XML"/>
>         <param name="context" value="default"/>
>         </span>
>   </boost_spans>
>
>   <analog_spans>
>         <span id="smg_prid">
>         <param name="dialplan" value="XML"/>
>         <param name="context" value="default"/>
>
>         </span>
>   </analog_spans>
> </configuration>
>
>
> And here is the lua code I'm using to dial out:
>
> sessiondata = "OpenZAP/smg_prid/<num>"
> new_session = freeswitch.Session(sessiondata)
>
>
> What am I missing here?
> Thanks,
> Neil
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From neilp at cs.stanford.edu  Sat May 15 15:26:52 2010
From: neilp at cs.stanford.edu (Neil Patel)
Date: Sat, 15 May 2010 15:26:52 -0700
Subject: [Freeswitch-users] span not defined error
In-Reply-To: <AANLkTin9wP0f74qEyA5GjtpVo7YacbS2C6VE9VnO31wj@mail.gmail.com>
References: <AANLkTik9DQcT5fpgcfb6KqFUZT4vXUuAjqTE5pFLxW-R@mail.gmail.com> 
	<AANLkTin9wP0f74qEyA5GjtpVo7YacbS2C6VE9VnO31wj@mail.gmail.com>
Message-ID: <AANLkTikleSqsMITY0lNf-Qyeg5kOb2ICJFDs4gO3UeZO@mail.gmail.com>

Span was originally in the boost section when I got this error, so I thought
I'd try it in analog and both. None work.

-Neil

On Sat, May 15, 2010 at 3:18 PM, Moises Silva <moises.silva at gmail.com>wrote:

> Why do you have 2 spans in openzap.conf.xml with the same name, in both the
> boost and analog sections?
> Moises Silva
> Senior Software Engineer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
> 9T3 Canada
> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>
>
> On Sat, May 15, 2010 at 3:52 PM, Neil Patel <neilp at cs.stanford.edu> wrote:
>
>> Hi All,
>>
>> I am trying to dial out over my PRI, and am getting this error:
>>
>> 2010-05-16 01:01:21.452392 [CRIT] zap_io.c:1139 SPAN NOT DEFINED!
>> 2010-05-16 01:01:21.452392 [ERR] mod_openzap.c:1154 No channels available
>> 2010-05-16 01:01:21.452392 [ERR] switch_ivr_originate.c:2249 Cannot create
>> outgoing channel of type [OpenZAP] cause: [NORMAL_CIRCUIT_CONGESTION]
>>
>>
>> This is my openzap.conf:
>>
>> [span wanpipe smg_prid]
>> name => smg_prid
>> trunk_type =>e1
>> b-channel => 1:1-15
>> b-channel => 1:17-31
>> trunk_type =>e1
>> b-channel => 2:1-15
>> b-channel => 2:17-31
>>
>>
>> This is my openzap.conf.xml:
>>
>> <configuration name="openzap.conf" description="OpenZAP Configuration">
>>   <settings>
>>     <param name="debug" value="0"/>
>>
>>   </settings>
>>   <boost_spans>
>>         <span id="smg_prid">
>>         <param name="dialplan" value="XML"/>
>>         <param name="context" value="default"/>
>>         </span>
>>   </boost_spans>
>>
>>   <analog_spans>
>>         <span id="smg_prid">
>>         <param name="dialplan" value="XML"/>
>>         <param name="context" value="default"/>
>>
>>         </span>
>>   </analog_spans>
>> </configuration>
>>
>>
>> And here is the lua code I'm using to dial out:
>>
>> sessiondata = "OpenZAP/smg_prid/<num>"
>> new_session = freeswitch.Session(sessiondata)
>>
>>
>> What am I missing here?
>> Thanks,
>> Neil
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at microcomaustralia.com.au  Sat May 15 18:03:30 2010
From: brian at microcomaustralia.com.au (Brian May)
Date: Sun, 16 May 2010 11:03:30 +1000
Subject: [Freeswitch-users] TDM400 & local echo
Message-ID: <AANLkTilDLf1aUvsr86JkRjkQZSnQR9NDTYXvgsbnvEeB@mail.gmail.com>

Hello,

When either I receive a phone call on my TDM400 card, everything I say
is echoed back to me, very loudly, with significant delay.

This doesn't seem to match the usual echo problems though, as like I
said the echo is only heard locally, not remotely. It seems to be with
different callers too, so it seems unlikely the echo is coming from
the caller.

Telephone calls that don't use the TDM400 card don't have this issue.

Any ideas?
-- 
Brian May <brian at microcomaustralia.com.au>


From jcasale at activenetwerx.com  Sat May 15 18:31:11 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Sun, 16 May 2010 01:31:11 +0000
Subject: [Freeswitch-users] TDM400 & local echo
In-Reply-To: <AANLkTilDLf1aUvsr86JkRjkQZSnQR9NDTYXvgsbnvEeB@mail.gmail.com>
References: <AANLkTilDLf1aUvsr86JkRjkQZSnQR9NDTYXvgsbnvEeB@mail.gmail.com>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E9A0AD99@prato.activenetwerx.local>

>Any ideas?

No, but I can confirm the *exact* behavior w/ my TDM400P card using
atrpms DAHDI rpms with Tzafrir Cohen's Oslec patches included.

If you do resolve this, I would be grateful to hear how!

jlc


From moises.silva at gmail.com  Sat May 15 22:20:33 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Sun, 16 May 2010 01:20:33 -0400
Subject: [Freeswitch-users] span not defined error
In-Reply-To: <AANLkTikleSqsMITY0lNf-Qyeg5kOb2ICJFDs4gO3UeZO@mail.gmail.com>
References: <AANLkTik9DQcT5fpgcfb6KqFUZT4vXUuAjqTE5pFLxW-R@mail.gmail.com>
	<AANLkTin9wP0f74qEyA5GjtpVo7YacbS2C6VE9VnO31wj@mail.gmail.com>
	<AANLkTikleSqsMITY0lNf-Qyeg5kOb2ICJFDs4gO3UeZO@mail.gmail.com>
Message-ID: <AANLkTikd_xI3QFUeWjJfVzNs5nc1tuK9_WHEu46Jsekd@mail.gmail.com>

Hi Again Neil,

I just noticed your dial string is incorrect. The correct syntax is:

OpenZAP/<span>/<chan_code>/[number]

The span and chan code are mandatory. The number is optional ( FXS channels
do not require a number, they just ring the FXO device connected to them).

The span is either a number ( span id, the id is a number assigned in the
order in which the span is defined in openzap.conf ) or a span name also as
specified in the [span wanpipe <span_name>] line in openzap.conf

The chan code is either a number ( for spans that support individual channel
selection, boost is NOT one of them ), or a channel hunting mode, there is
currently 2 modes, "a" is top down and "A" is bottom up.

So, this is a valid string for you case:

OpenZAP/smg_prid/a/<number>

In the specific case of boost in socket mode ( openzap only supports socket
mode ) the number may contain @gX where X is a group ( for hunting as
configured in /etc/wanpipe/smg_pri.conf). Boost signaling are a special case
because the hunting for channels is not done within FreeSWITCH but in
sangoma_prid binary ( in the new OpenZAP version called FreeTDM this has
changed depending on configuration).

Bottom line, this should work:

OpenZAP/smg_prid/a/1234 at g1

If you have g1 configured in /etc/wanpipe/smg_pri.conf

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com


On Sat, May 15, 2010 at 6:26 PM, Neil Patel <neilp at cs.stanford.edu> wrote:

> Span was originally in the boost section when I got this error, so I
> thought I'd try it in analog and both. None work.
>
> -Neil
>
> On Sat, May 15, 2010 at 3:18 PM, Moises Silva <moises.silva at gmail.com>wrote:
>
>> Why do you have 2 spans in openzap.conf.xml with the same name, in both
>> the boost and analog sections?
>> Moises Silva
>> Senior Software Engineer
>> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
>> 9T3 Canada
>> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>>
>>
>> On Sat, May 15, 2010 at 3:52 PM, Neil Patel <neilp at cs.stanford.edu>wrote:
>>
>>> Hi All,
>>>
>>> I am trying to dial out over my PRI, and am getting this error:
>>>
>>> 2010-05-16 01:01:21.452392 [CRIT] zap_io.c:1139 SPAN NOT DEFINED!
>>> 2010-05-16 01:01:21.452392 [ERR] mod_openzap.c:1154 No channels available
>>> 2010-05-16 01:01:21.452392 [ERR] switch_ivr_originate.c:2249 Cannot
>>> create outgoing channel of type [OpenZAP] cause: [NORMAL_CIRCUIT_CONGESTION]
>>>
>>>
>>> This is my openzap.conf:
>>>
>>> [span wanpipe smg_prid]
>>> name => smg_prid
>>> trunk_type =>e1
>>> b-channel => 1:1-15
>>> b-channel => 1:17-31
>>> trunk_type =>e1
>>> b-channel => 2:1-15
>>> b-channel => 2:17-31
>>>
>>>
>>> This is my openzap.conf.xml:
>>>
>>> <configuration name="openzap.conf" description="OpenZAP Configuration">
>>>   <settings>
>>>     <param name="debug" value="0"/>
>>>
>>>   </settings>
>>>   <boost_spans>
>>>         <span id="smg_prid">
>>>         <param name="dialplan" value="XML"/>
>>>         <param name="context" value="default"/>
>>>         </span>
>>>   </boost_spans>
>>>
>>>   <analog_spans>
>>>         <span id="smg_prid">
>>>         <param name="dialplan" value="XML"/>
>>>         <param name="context" value="default"/>
>>>
>>>         </span>
>>>   </analog_spans>
>>> </configuration>
>>>
>>>
>>> And here is the lua code I'm using to dial out:
>>>
>>> sessiondata = "OpenZAP/smg_prid/<num>"
>>> new_session = freeswitch.Session(sessiondata)
>>>
>>>
>>> What am I missing here?
>>> Thanks,
>>> Neil
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From babak.freeswitch at gmail.com  Sun May 16 02:07:59 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Sun, 16 May 2010 13:37:59 +0430
Subject: [Freeswitch-users] EventConsumer and multi-threading
Message-ID: <AANLkTinz4WHJrO_6q3anFdtiU3wZyaH8XHaaGFjVpQhB@mail.gmail.com>

Hi
I'm using mod_managed and I've used EventConsumer on a thread to listen for
incomming events but whenever I wanna reload the .dll file of my script it
complains about thread aborting. I used a dummy custom:dead event to end the
thread but next time my dll loaded the custom:dead event is still in events
and the tread ends. Is there any workaround to flush EventConsumer??
Thanks
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From bwibowo at gmail.com  Sun May 16 06:46:50 2010
From: bwibowo at gmail.com (budi wibowo)
Date: Sun, 16 May 2010 20:46:50 +0700
Subject: [Freeswitch-users] mod_ivr with condition
Message-ID: <AANLkTilT9qs9b61U6w4thmO3KPfMh3PM0VqZRm_l4ryi@mail.gmail.com>

dear all
i setup ivr successfully following xml sample, i want to add functionality
to give condition based on calling number,
generally i want this:
if calling_id =1000
 then do someaction
 else do some other action


can i use xml for this or should use javascript ?

regards

budi
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From bwibowo at gmail.com  Sun May 16 08:34:27 2010
From: bwibowo at gmail.com (budi wibowo)
Date: Sun, 16 May 2010 22:34:27 +0700
Subject: [Freeswitch-users] resources path mapping
Message-ID: <AANLkTil8IZKV16u1IJfJFf-ityXlr9HlVgYP4TG7GL44@mail.gmail.com>

dear all
just check /usr/local/freeswitch/conf/lang/en/demo/demo-ivr.xml
and i found
<!-- Menu option 9: Repeat these options -->
        <action function="play-file"
data="ivr/ivr-to_repeat_these_options.wav"/>
        <action function="play-file" data="ivr/ivr-please.wav"/>
        <action function="play-file" data="voicemail/vm-press.wav"/>
        <action function="play-file" data="digits/9.wav"/>

digits/9.wav actually
read /usr/local/freeswitch/sounds/en/us/callie/digits/8000/9.wav
ivr/ivr-please.wav actually
read /usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-please.wav

where's the mapping definition for 2  things above?


regards

budi wibowo
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From shroukkhan at softverk.is  Sun May 16 09:32:36 2010
From: shroukkhan at softverk.is (Shrouk Khan)
Date: Sun, 16 May 2010 23:32:36 +0700
Subject: [Freeswitch-users] Configuring jbilling with freeswitch
Message-ID: <AANLkTikLDTtEhbL8ji81_m6EQLG9f0zC0vW90iaraMof@mail.gmail.com>

hi , has anyone ever been able to use jbilling with freeswitch ? i have both
of them installed , but can not figure where to start ? is there any
guidlines or anything that anyone knows of?

-- 
Regards

Shrouk Khan (Khan)
System Administrator / Telecommunication System Developer
Office:   +354 4400807 (Reykjavik)
          +44   2031370800 (London)
Mobile:  +66  875049439 (Bangkok)

Web: www.softverk.is

Reykjavik, Iceland  //  London, UK   //  Bangkok, Thailand
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From lands at freenet.de  Sun May 16 10:28:14 2010
From: lands at freenet.de (lands at freenet.de)
Date: Sun, 16 May 2010 17:28:14 -0000
Subject: [Freeswitch-users] headset for CELT fullband conferencing
Message-ID: <4C2CD024.10403@freenet.de>

Hi,

I?m looking for the "best" headset for CELT fullband conferencing.
In an old thread the Logitech 350 premium headset was suggested. The
Logitech seem to be "only" wideband.

What do you use nowadays?


From garrison at codefix.net  Sun May 16 14:42:24 2010
From: garrison at codefix.net (Garrison Hoffman)
Date: Sun, 16 May 2010 17:42:24 -0400
Subject: [Freeswitch-users] per domain vm templates
Message-ID: <1274046144.29140.37.camel@strontium>

Does mod_voicemail support per domain or per user templates? I'd like to
do something like:

<param name="web-template-file"
value="templates/$${voicemail_domain}/web-vm.tpl"/>

I've tried a couple variations with no success, is there a way to
accomplish this without preprocessor variables?

I guess plan 'B' would be to come at this via XML-RPC but that seems a
bit like reinventing the wheel.

-gh



From brian at freeswitch.org  Sun May 16 15:39:17 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 16 May 2010 17:39:17 -0500
Subject: [Freeswitch-users] per domain vm templates
In-Reply-To: <1274046144.29140.37.camel@strontium>
References: <1274046144.29140.37.camel@strontium>
Message-ID: <C5B2D416-B7CE-4F72-9865-0FD0CAA10E74@freeswitch.org>

not with $$ you should consider doing XML curl the you can do what ever like.

Sent from my iPad

On May 16, 2010, at 4:42 PM, Garrison Hoffman <garrison at codefix.net> wrote:

> Does mod_voicemail support per domain or per user templates? I'd like to
> do something like:
> 
> <param name="web-template-file"
> value="templates/$${voicemail_domain}/web-vm.tpl"/>
> 
> I've tried a couple variations with no success, is there a way to
> accomplish this without preprocessor variables?
> 
> I guess plan 'B' would be to come at this via XML-RPC but that seems a
> bit like reinventing the wheel.
> 
> -gh
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From mike at jerris.com  Sun May 16 16:43:03 2010
From: mike at jerris.com (Michael Jerris)
Date: Sun, 16 May 2010 19:43:03 -0400
Subject: [Freeswitch-users] resources path mapping
In-Reply-To: <AANLkTil8IZKV16u1IJfJFf-ityXlr9HlVgYP4TG7GL44@mail.gmail.com>
References: <AANLkTil8IZKV16u1IJfJFf-ityXlr9HlVgYP4TG7GL44@mail.gmail.com>
Message-ID: <8861F62D-A5DD-4415-AA09-1906DDF2E4AB@jerris.com>

sound_prefix 

On May 16, 2010, at 11:34 AM, budi wibowo wrote:

> dear all 
> just check /usr/local/freeswitch/conf/lang/en/demo/demo-ivr.xml
> and i found 
> <!-- Menu option 9: Repeat these options -->
>         <action function="play-file" data="ivr/ivr-to_repeat_these_options.wav"/>
>         <action function="play-file" data="ivr/ivr-please.wav"/>
>         <action function="play-file" data="voicemail/vm-press.wav"/>
>         <action function="play-file" data="digits/9.wav"/>
> 
> digits/9.wav actually read /usr/local/freeswitch/sounds/en/us/callie/digits/8000/9.wav 
> ivr/ivr-please.wav actually read /usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-please.wav
> 
> where's the mapping definition for 2  things above?
> 



From neilp at cs.stanford.edu  Sun May 16 17:16:42 2010
From: neilp at cs.stanford.edu (Neil Patel)
Date: Sun, 16 May 2010 17:16:42 -0700
Subject: [Freeswitch-users] span not defined error
In-Reply-To: <AANLkTikd_xI3QFUeWjJfVzNs5nc1tuK9_WHEu46Jsekd@mail.gmail.com>
References: <AANLkTik9DQcT5fpgcfb6KqFUZT4vXUuAjqTE5pFLxW-R@mail.gmail.com> 
	<AANLkTin9wP0f74qEyA5GjtpVo7YacbS2C6VE9VnO31wj@mail.gmail.com> 
	<AANLkTikleSqsMITY0lNf-Qyeg5kOb2ICJFDs4gO3UeZO@mail.gmail.com> 
	<AANLkTikd_xI3QFUeWjJfVzNs5nc1tuK9_WHEu46Jsekd@mail.gmail.com>
Message-ID: <AANLkTinbeTgmLDAcxyF3B_JqO7J59XFTbTOtLio0ByaM@mail.gmail.com>

Thanks Moises, this is very helpful!

After fixing the dialstring, I am still getting the following error from FS:

> originate openzap/smg_prid/a/<num> &echo
-ERR NORMAL_CIRCUIT_CONGESTION

2010-05-17 05:39:02.883264 [WARNING] ozmod_sangoma_boost.c:346 TX EVENT:
CALL_START:(80) [w1g1] CSid=1 Seq=0 Cn=[FreeSWITCH] Cd=[9428500597]
Ci=[0000000000] Rdnis=[]
freeswitch at internal> 2010-05-17 05:39:02.955336 [WARNING]
ozmod_sangoma_boost.c:1312 RX EVENT (N): CALL_START_NACK:(82) [w256g256]
Rc=0 CSid=1 Seq=1
2010-05-17 05:39:02.955336 [WARNING] sangoma_boost_client.c:220 TX EVENT
(N): CALL_START_NACK_ACK:(83) [w1g1] Rc=0 CSid=1 Seq=1
2010-05-17 05:39:02.956586 [ERR] mod_openzap.c:1154 No channels available
2010-05-17 05:39:02.956586 [ERR] switch_ivr_originate.c:2249 Cannot create
outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]



Here is the relevant logging from /var/log/sangoma_mgd.log:

May 17 05:39:02 otalo sangoma_prid: Rx SMG [CALL_START s:0 c:0 id:1]
May 17 05:39:02 otalo sangoma_prid: g1:Outgoing call (Smg-ID:1)
May 17 05:39:02 otalo sangoma_prid: s1:Outgoing call ChanRq:1
Called-Nb[9428500597] Calling-Nb[0000000000] (Smg-ID:1)
May 17 05:39:02 otalo sangoma_prid: Rx Tsoft [7:StatusIn id:0]
May 17 05:39:02 otalo sangoma_prid: s1:Received cause-ind-An IE or parameter
does not exist(99)
May 17 05:39:02 otalo sangoma_prid: Rx Tsoft [2:DiscIn id:0]
May 17 05:39:02 otalo sangoma_prid: s1c1:Remote released-Unknown(0)
May 17 05:39:02 otalo sangoma_prid: Rx SMG [CALL_START_NACK_ACK s:0 c:0
id:1]
May 17 05:39:02 otalo sangoma_prid: g1:Call cleared (SMG-ID:1)


I updated /etc/wanpipe/smg_pri.conf to have group=1 uncommented.

Please note that my box is in India, making calls over a PRI line set up
there. I also changed my tones.conf to match what's on the
wiki<http://wiki.freeswitch.org/wiki/Tones.conf_Example>for India.

Thanks,
Neil

On Sat, May 15, 2010 at 10:20 PM, Moises Silva <moises.silva at gmail.com>wrote:

> Hi Again Neil,
>
> I just noticed your dial string is incorrect. The correct syntax is:
>
> OpenZAP/<span>/<chan_code>/[number]
>
> The span and chan code are mandatory. The number is optional ( FXS channels
> do not require a number, they just ring the FXO device connected to them).
>
> The span is either a number ( span id, the id is a number assigned in the
> order in which the span is defined in openzap.conf ) or a span name also as
> specified in the [span wanpipe <span_name>] line in openzap.conf
>
> The chan code is either a number ( for spans that support individual
> channel selection, boost is NOT one of them ), or a channel hunting mode,
> there is currently 2 modes, "a" is top down and "A" is bottom up.
>
> So, this is a valid string for you case:
>
> OpenZAP/smg_prid/a/<number>
>
> In the specific case of boost in socket mode ( openzap only supports socket
> mode ) the number may contain @gX where X is a group ( for hunting as
> configured in /etc/wanpipe/smg_pri.conf). Boost signaling are a special case
> because the hunting for channels is not done within FreeSWITCH but in
> sangoma_prid binary ( in the new OpenZAP version called FreeTDM this has
> changed depending on configuration).
>
> Bottom line, this should work:
>
> OpenZAP/smg_prid/a/1234 at g1
>
> If you have g1 configured in /etc/wanpipe/smg_pri.conf
>
> Moises Silva
> Senior Software Engineer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
> 9T3 Canada
> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>
>
> On Sat, May 15, 2010 at 6:26 PM, Neil Patel <neilp at cs.stanford.edu> wrote:
>
>> Span was originally in the boost section when I got this error, so I
>> thought I'd try it in analog and both. None work.
>>
>> -Neil
>>
>> On Sat, May 15, 2010 at 3:18 PM, Moises Silva <moises.silva at gmail.com>wrote:
>>
>>> Why do you have 2 spans in openzap.conf.xml with the same name, in both
>>> the boost and analog sections?
>>> Moises Silva
>>> Senior Software Engineer
>>> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
>>> 9T3 Canada
>>> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>>>
>>>
>>> On Sat, May 15, 2010 at 3:52 PM, Neil Patel <neilp at cs.stanford.edu>wrote:
>>>
>>>> Hi All,
>>>>
>>>> I am trying to dial out over my PRI, and am getting this error:
>>>>
>>>> 2010-05-16 01:01:21.452392 [CRIT] zap_io.c:1139 SPAN NOT DEFINED!
>>>> 2010-05-16 01:01:21.452392 [ERR] mod_openzap.c:1154 No channels
>>>> available
>>>> 2010-05-16 01:01:21.452392 [ERR] switch_ivr_originate.c:2249 Cannot
>>>> create outgoing channel of type [OpenZAP] cause: [NORMAL_CIRCUIT_CONGESTION]
>>>>
>>>>
>>>> This is my openzap.conf:
>>>>
>>>> [span wanpipe smg_prid]
>>>> name => smg_prid
>>>> trunk_type =>e1
>>>> b-channel => 1:1-15
>>>> b-channel => 1:17-31
>>>> trunk_type =>e1
>>>> b-channel => 2:1-15
>>>> b-channel => 2:17-31
>>>>
>>>>
>>>> This is my openzap.conf.xml:
>>>>
>>>> <configuration name="openzap.conf" description="OpenZAP Configuration">
>>>>   <settings>
>>>>     <param name="debug" value="0"/>
>>>>
>>>>   </settings>
>>>>   <boost_spans>
>>>>         <span id="smg_prid">
>>>>         <param name="dialplan" value="XML"/>
>>>>         <param name="context" value="default"/>
>>>>         </span>
>>>>   </boost_spans>
>>>>
>>>>   <analog_spans>
>>>>         <span id="smg_prid">
>>>>         <param name="dialplan" value="XML"/>
>>>>         <param name="context" value="default"/>
>>>>
>>>>         </span>
>>>>   </analog_spans>
>>>> </configuration>
>>>>
>>>>
>>>> And here is the lua code I'm using to dial out:
>>>>
>>>> sessiondata = "OpenZAP/smg_prid/<num>"
>>>> new_session = freeswitch.Session(sessiondata)
>>>>
>>>>
>>>> What am I missing here?
>>>> Thanks,
>>>> Neil
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lloyd.aloysius at sunteltech.ca  Sun May 16 21:28:40 2010
From: lloyd.aloysius at sunteltech.ca (Aloysius Lloyd)
Date: Mon, 17 May 2010 00:28:40 -0400
Subject: [Freeswitch-users] Gateway Registration Issues
In-Reply-To: <983B7F89-A86E-4AFE-91E6-553D1F0B6495@gmail.com>
References: <AANLkTim-d2_nD3lLS3YlqciwrgJpGP5jMyCfSWtxI_Ly@mail.gmail.com> 
	<983B7F89-A86E-4AFE-91E6-553D1F0B6495@gmail.com>
Message-ID: <AANLkTinmN69iWE8S3BsdgcF9Slwkgktf4u6ZP32p1rmv@mail.gmail.com>

Hi David,

Thank you for the answer. your suggestion solve my problem.

Thanks,
Lloyd


On Thu, May 13, 2010 at 8:58 AM, David Ponzone <david.ponzone at gmail.com>wrote:

> Aloysius,,
>
> That is an old-skool carrier.
> You have to add this line to your gateway params:
>
> <param name="extension-in-contact" value="true"/>
>
>
>  David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 13/05/2010 ? 14:46, Aloysius Lloyd a ?crit :
>
> Hi All,
>
> I am trying to connect IRISTEL ( CLEC ) in Canada and FreeSWITCH. I have
> the following gateway configuration
>
> <include>
>   <gateway name="wwc_test">
>     <param name="auth-username" value="username"/>
>     <param name="username" value="14161231234" />
>     <param name="password" value="password" />
>     <param name="proxy" value="sbc1.iristel.net" />
>     <param name="realm" value="iristel.net" />
>     <param name="expire-seconds" value="120"/>
>     <param name="register" value="true"/>
>     <param name="retry-seconds" value="30"/>
>     <param name="ping" value="25"/>
>   </gateway>
> </include>
>
>
> The above configuration is not working. I have the following error in cli
> Error.
>
> *2010-05-13 08:34:15.987465 [ERR] sofia_reg.c:1501 14161231234
> Registration Failed with status Conflict [409]. failure #1*
>
> ====
>
> Also carrier saying there is conflict in the contact field.Here is how
> they receive now.
>
> *Contact: <
> sip:gw+test_iristel at 10.20.30.1:5080;transport=udp;gw=test_iristel>.*
>
> But the contact field should be like below
>
> Contact: <14161231234 at 10.20.30.1:5080;
>
> Any suggestions?
>
> Thanks in advance.
>
> Lloyd
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tculjaga at gmail.com  Mon May 17 01:14:16 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 17 May 2010 10:14:16 +0200
Subject: [Freeswitch-users] fetch caller profile variables
In-Reply-To: <AANLkTim4KO8XRwl-n9-tq-BSFpwC2Jds1H752Rh9uNIk@mail.gmail.com>
References: <AANLkTinXlIUddb_xAWh1u3DLpAuuWQhisOOA8ygWLGnU@mail.gmail.com>
	<AANLkTim4KO8XRwl-n9-tq-BSFpwC2Jds1H752Rh9uNIk@mail.gmail.com>
Message-ID: <AANLkTimnITwNVY-jlgsxYE4fYQCOWnQpu6eR2a026P0H@mail.gmail.com>

On Thu, May 13, 2010 at 4:39 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> answer_stamp is a regular channel variable not a caller profile variable
> ${answer_stamp} is correct but its only set once the channel is hungup
>
> billing is best done in a separate process from the dialplan on the backend
> via EVENTS or CDR
>
>
so, when/how do i set these variables to have them available in the CDR ?

T.
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From leo.zibi at gmail.com  Mon May 17 01:46:01 2010
From: leo.zibi at gmail.com (leo.zibi at gmail.com)
Date: Mon, 17 May 2010 10:46:01 +0200
Subject: [Freeswitch-users] fetch caller profile variables
In-Reply-To: <AANLkTimnITwNVY-jlgsxYE4fYQCOWnQpu6eR2a026P0H@mail.gmail.com>
References: <AANLkTinXlIUddb_xAWh1u3DLpAuuWQhisOOA8ygWLGnU@mail.gmail.com>	<AANLkTim4KO8XRwl-n9-tq-BSFpwC2Jds1H752Rh9uNIk@mail.gmail.com>
	<AANLkTimnITwNVY-jlgsxYE4fYQCOWnQpu6eR2a026P0H@mail.gmail.com>
Message-ID: <4BF10249.8010105@gmail.com>

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From miconda at gmail.com  Mon May 17 02:01:14 2010
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Mon, 17 May 2010 11:01:14 +0200
Subject: [Freeswitch-users] kamailio and freeswitch integration
Message-ID: <4BF105DA.2060209@gmail.com>

Hello,

I put together a tutorial about using kamailio (openser) and freeswtich 
together: the proxy takes care of authentication and registration, 
freeswitch of media services, here is a link:

http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms

Is it ok to upload it to FS wiki so others can add to it?

Cheers,
Daniel

-- 
Daniel-Constantin Mierla
Kamailio (OpenSER) Advanced Training
Miami, Fl, USA - June 21-23, 2010
http://www.asipto.com/index.php/kamailio-advanced-training/



From tculjaga at gmail.com  Mon May 17 02:19:55 2010
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 17 May 2010 11:19:55 +0200
Subject: [Freeswitch-users] fetch caller profile variables
In-Reply-To: <4BF10249.8010105@gmail.com>
References: <AANLkTinXlIUddb_xAWh1u3DLpAuuWQhisOOA8ygWLGnU@mail.gmail.com>
	<AANLkTim4KO8XRwl-n9-tq-BSFpwC2Jds1H752Rh9uNIk@mail.gmail.com>
	<AANLkTimnITwNVY-jlgsxYE4fYQCOWnQpu6eR2a026P0H@mail.gmail.com>
	<4BF10249.8010105@gmail.com>
Message-ID: <AANLkTimMWUadGvAaL-sMUvVNTQxiF2AjbrNDGliu6MJU@mail.gmail.com>

> On Thu, May 13, 2010 at 4:39 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> answer_stamp is a regular channel variable not a caller profile variable
>> ${answer_stamp} is correct but its only set once the channel is hungup
>>
>>  billing is best done in a separate process from the dialplan on the
>> backend via EVENTS or CDR
>>
>>
> so, when/how do i set these variables to have them available in the CDR ?
>
> T.
>
>         Dialplan
>
>      <action application="export" data="billing_model_id=0"/>
>       <action application="export" data="country_name=UE"/>
>       <action application="export" data="area_name=918"/>
>       <action application="export" data="zone_name=blah"/>
>
>
thanks, and when do i execute export ?
come vars are being populated on hangup...




> --
>
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From leo.zibi at gmail.com  Mon May 17 03:35:38 2010
From: leo.zibi at gmail.com (leo.zibi at gmail.com)
Date: Mon, 17 May 2010 12:35:38 +0200
Subject: [Freeswitch-users] fetch caller profile variables
In-Reply-To: <AANLkTimMWUadGvAaL-sMUvVNTQxiF2AjbrNDGliu6MJU@mail.gmail.com>
References: <AANLkTinXlIUddb_xAWh1u3DLpAuuWQhisOOA8ygWLGnU@mail.gmail.com>	<AANLkTim4KO8XRwl-n9-tq-BSFpwC2Jds1H752Rh9uNIk@mail.gmail.com>	<AANLkTimnITwNVY-jlgsxYE4fYQCOWnQpu6eR2a026P0H@mail.gmail.com>	<4BF10249.8010105@gmail.com>
	<AANLkTimMWUadGvAaL-sMUvVNTQxiF2AjbrNDGliu6MJU@mail.gmail.com>
Message-ID: <4BF11BFA.6060406@gmail.com>

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From bobc at devassert.com  Sun May 16 21:55:03 2010
From: bobc at devassert.com (Bob Coleman)
Date: Mon, 17 May 2010 16:55:03 +1200
Subject: [Freeswitch-users] Setup help for outbound dialing
In-Reply-To: <AANLkTikGzzmdhr8iHKy39zGdvcNPpsKkBH3snt_6cfDW@mail.gmail.com>
References: <AANLkTikGzzmdhr8iHKy39zGdvcNPpsKkBH3snt_6cfDW@mail.gmail.com>
Message-ID: <AANLkTinjlIr97zIhXgS0LGmjfNOeXWvNObS4gweZZnd_@mail.gmail.com>

Hi,

My setup is basically as follows:

Have a sip profile that allows incoming calls(lets call it internal).
Have setup dialplans to accept the inbound ddi's that I want to allow
routing to an ivr application.

Our server is hosted inside a sip providers network, so we dont use
gateways etc, direct sip trunk via internal sofia profile.

I am struggling to setup outbound calling. I can make a call through a
gateway, but am not sure what to setup to allow an outbound call
through our internal profile.

eg originate sofia/internal/<phonenumber>@<ipaddress> <extension>

Any help would be appreciated. Not sure what other info you would like
but just ask

Thanks

Bob


From tomb at cachecomm.com  Mon May 17 07:18:48 2010
From: tomb at cachecomm.com (Tom)
Date: Mon, 17 May 2010 08:18:48 -0600
Subject: [Freeswitch-users] span not defined error
In-Reply-To: <AANLkTik9DQcT5fpgcfb6KqFUZT4vXUuAjqTE5pFLxW-R@mail.gmail.com>
References: <AANLkTik9DQcT5fpgcfb6KqFUZT4vXUuAjqTE5pFLxW-R@mail.gmail.com>
Message-ID: <4BF15048.5090002@cachecomm.com>

Neil Patel wrote:
> Hi All,
>
> I am trying to dial out over my PRI, and am getting this error:
>
> 2010-05-16 01:01:21.452392 [CRIT] zap_io.c:1139 SPAN NOT DEFINED!
> 2010-05-16 01:01:21.452392 [ERR] mod_openzap.c:1154 No channels available
> 2010-05-16 01:01:21.452392 [ERR] switch_ivr_originate.c:2249 Cannot 
> create outgoing channel of type [OpenZAP] cause: 
> [NORMAL_CIRCUIT_CONGESTION]
>
>
> This is my openzap.conf:
>
>     [span wanpipe smg_prid]
>     name => smg_prid
>     trunk_type =>e1
>     b-channel => 1:1-15
>     b-channel => 1:17-31
>     trunk_type =>e1
>     b-channel => 2:1-15
>     b-channel => 2:17-31
>
>
> This is my openzap.conf.xml:
>
>     <configuration name="openzap.conf" description="OpenZAP
>     Configuration">
>       <settings>
>         <param name="debug" value="0"/>
>        
>       </settings>
>       <boost_spans>
>             <span id="smg_prid">
>             <param name="dialplan" value="XML"/>
>             <param name="context" value="default"/>
>             </span>
>       </boost_spans>
>       
>       <analog_spans>
>             <span id="smg_prid">
>             <param name="dialplan" value="XML"/>
>             <param name="context" value="default"/>
>           
>             </span>
>       </analog_spans>
>     </configuration>
>
>
> And here is the lua code I'm using to dial out:
>
>     sessiondata = "OpenZAP/smg_prid/<num>"
>     new_session = freeswitch.Session(sessiondata)
>
>
> What am I missing here? 
> Thanks,
> Neil
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   
Neil,
Which card are you using?

Tom


From moises.silva at gmail.com  Mon May 17 07:44:41 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Mon, 17 May 2010 10:44:41 -0400
Subject: [Freeswitch-users] span not defined error
In-Reply-To: <AANLkTinbeTgmLDAcxyF3B_JqO7J59XFTbTOtLio0ByaM@mail.gmail.com>
References: <AANLkTik9DQcT5fpgcfb6KqFUZT4vXUuAjqTE5pFLxW-R@mail.gmail.com>
	<AANLkTin9wP0f74qEyA5GjtpVo7YacbS2C6VE9VnO31wj@mail.gmail.com>
	<AANLkTikleSqsMITY0lNf-Qyeg5kOb2ICJFDs4gO3UeZO@mail.gmail.com>
	<AANLkTikd_xI3QFUeWjJfVzNs5nc1tuK9_WHEu46Jsekd@mail.gmail.com>
	<AANLkTinbeTgmLDAcxyF3B_JqO7J59XFTbTOtLio0ByaM@mail.gmail.com>
Message-ID: <AANLkTim6OwUdrXQsdsSo9npyMVaezHlVLfhZfNGszEQP@mail.gmail.com>

Try taking a protocol dump using wanpipemon on the d-channel so we can see
the full details of the call setup.

http://wiki.sangoma.com/wanpipe-wireshark-pcap-pri-bri-wan-t1-e1-tracing#pri_bri_wireshark

Then paste a link here to the pcap file somewhere.

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com


On Sun, May 16, 2010 at 8:16 PM, Neil Patel <neilp at cs.stanford.edu> wrote:

> Thanks Moises, this is very helpful!
>
> After fixing the dialstring, I am still getting the following error from
> FS:
>
> > originate openzap/smg_prid/a/<num> &echo
> -ERR NORMAL_CIRCUIT_CONGESTION
>
> 2010-05-17 05:39:02.883264 [WARNING] ozmod_sangoma_boost.c:346 TX EVENT:
> CALL_START:(80) [w1g1] CSid=1 Seq=0 Cn=[FreeSWITCH] Cd=[9428500597]
> Ci=[0000000000] Rdnis=[]
> freeswitch at internal> 2010-05-17 05:39:02.955336 [WARNING]
> ozmod_sangoma_boost.c:1312 RX EVENT (N): CALL_START_NACK:(82) [w256g256]
> Rc=0 CSid=1 Seq=1
> 2010-05-17 05:39:02.955336 [WARNING] sangoma_boost_client.c:220 TX EVENT
> (N): CALL_START_NACK_ACK:(83) [w1g1] Rc=0 CSid=1 Seq=1
> 2010-05-17 05:39:02.956586 [ERR] mod_openzap.c:1154 No channels available
> 2010-05-17 05:39:02.956586 [ERR] switch_ivr_originate.c:2249 Cannot create
> outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
>
>
>
> Here is the relevant logging from /var/log/sangoma_mgd.log:
>
> May 17 05:39:02 otalo sangoma_prid: Rx SMG [CALL_START s:0 c:0 id:1]
> May 17 05:39:02 otalo sangoma_prid: g1:Outgoing call (Smg-ID:1)
> May 17 05:39:02 otalo sangoma_prid: s1:Outgoing call ChanRq:1
> Called-Nb[9428500597] Calling-Nb[0000000000] (Smg-ID:1)
> May 17 05:39:02 otalo sangoma_prid: Rx Tsoft [7:StatusIn id:0]
> May 17 05:39:02 otalo sangoma_prid: s1:Received cause-ind-An IE or
> parameter does not exist(99)
> May 17 05:39:02 otalo sangoma_prid: Rx Tsoft [2:DiscIn id:0]
> May 17 05:39:02 otalo sangoma_prid: s1c1:Remote released-Unknown(0)
> May 17 05:39:02 otalo sangoma_prid: Rx SMG [CALL_START_NACK_ACK s:0 c:0
> id:1]
> May 17 05:39:02 otalo sangoma_prid: g1:Call cleared (SMG-ID:1)
>
>
> I updated /etc/wanpipe/smg_pri.conf to have group=1 uncommented.
>
> Please note that my box is in India, making calls over a PRI line set up
> there. I also changed my tones.conf to match what's on the wiki<http://wiki.freeswitch.org/wiki/Tones.conf_Example>for India.
>
> Thanks,
> Neil
>
> On Sat, May 15, 2010 at 10:20 PM, Moises Silva <moises.silva at gmail.com>wrote:
>
>> Hi Again Neil,
>>
>> I just noticed your dial string is incorrect. The correct syntax is:
>>
>> OpenZAP/<span>/<chan_code>/[number]
>>
>> The span and chan code are mandatory. The number is optional ( FXS
>> channels do not require a number, they just ring the FXO device connected to
>> them).
>>
>> The span is either a number ( span id, the id is a number assigned in the
>> order in which the span is defined in openzap.conf ) or a span name also as
>> specified in the [span wanpipe <span_name>] line in openzap.conf
>>
>> The chan code is either a number ( for spans that support individual
>> channel selection, boost is NOT one of them ), or a channel hunting mode,
>> there is currently 2 modes, "a" is top down and "A" is bottom up.
>>
>> So, this is a valid string for you case:
>>
>> OpenZAP/smg_prid/a/<number>
>>
>> In the specific case of boost in socket mode ( openzap only supports
>> socket mode ) the number may contain @gX where X is a group ( for hunting as
>> configured in /etc/wanpipe/smg_pri.conf). Boost signaling are a special case
>> because the hunting for channels is not done within FreeSWITCH but in
>> sangoma_prid binary ( in the new OpenZAP version called FreeTDM this has
>> changed depending on configuration).
>>
>> Bottom line, this should work:
>>
>> OpenZAP/smg_prid/a/1234 at g1
>>
>> If you have g1 configured in /etc/wanpipe/smg_pri.conf
>>
>> Moises Silva
>> Senior Software Engineer
>> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
>> 9T3 Canada
>> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>>
>>
>> On Sat, May 15, 2010 at 6:26 PM, Neil Patel <neilp at cs.stanford.edu>wrote:
>>
>>> Span was originally in the boost section when I got this error, so I
>>> thought I'd try it in analog and both. None work.
>>>
>>> -Neil
>>>
>>> On Sat, May 15, 2010 at 3:18 PM, Moises Silva <moises.silva at gmail.com>wrote:
>>>
>>>> Why do you have 2 spans in openzap.conf.xml with the same name, in both
>>>> the boost and analog sections?
>>>> Moises Silva
>>>> Senior Software Engineer
>>>> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
>>>> 9T3 Canada
>>>> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>>>>
>>>>
>>>> On Sat, May 15, 2010 at 3:52 PM, Neil Patel <neilp at cs.stanford.edu>wrote:
>>>>
>>>>> Hi All,
>>>>>
>>>>> I am trying to dial out over my PRI, and am getting this error:
>>>>>
>>>>> 2010-05-16 01:01:21.452392 [CRIT] zap_io.c:1139 SPAN NOT DEFINED!
>>>>> 2010-05-16 01:01:21.452392 [ERR] mod_openzap.c:1154 No channels
>>>>> available
>>>>> 2010-05-16 01:01:21.452392 [ERR] switch_ivr_originate.c:2249 Cannot
>>>>> create outgoing channel of type [OpenZAP] cause: [NORMAL_CIRCUIT_CONGESTION]
>>>>>
>>>>>
>>>>> This is my openzap.conf:
>>>>>
>>>>> [span wanpipe smg_prid]
>>>>> name => smg_prid
>>>>> trunk_type =>e1
>>>>> b-channel => 1:1-15
>>>>> b-channel => 1:17-31
>>>>> trunk_type =>e1
>>>>> b-channel => 2:1-15
>>>>> b-channel => 2:17-31
>>>>>
>>>>>
>>>>> This is my openzap.conf.xml:
>>>>>
>>>>> <configuration name="openzap.conf" description="OpenZAP Configuration">
>>>>>   <settings>
>>>>>     <param name="debug" value="0"/>
>>>>>
>>>>>   </settings>
>>>>>   <boost_spans>
>>>>>         <span id="smg_prid">
>>>>>         <param name="dialplan" value="XML"/>
>>>>>         <param name="context" value="default"/>
>>>>>         </span>
>>>>>   </boost_spans>
>>>>>
>>>>>   <analog_spans>
>>>>>         <span id="smg_prid">
>>>>>         <param name="dialplan" value="XML"/>
>>>>>         <param name="context" value="default"/>
>>>>>
>>>>>         </span>
>>>>>   </analog_spans>
>>>>> </configuration>
>>>>>
>>>>>
>>>>> And here is the lua code I'm using to dial out:
>>>>>
>>>>> sessiondata = "OpenZAP/smg_prid/<num>"
>>>>> new_session = freeswitch.Session(sessiondata)
>>>>>
>>>>>
>>>>> What am I missing here?
>>>>> Thanks,
>>>>> Neil
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon May 17 07:54:51 2010
From: msc at freeswitch.org (Michael S Collins)
Date: Mon, 17 May 2010 07:54:51 -0700
Subject: [Freeswitch-users] kamailio and freeswitch integration
In-Reply-To: <4BF105DA.2060209@gmail.com>
References: <4BF105DA.2060209@gmail.com>
Message-ID: <3BA4F163-24A3-4C91-A3B7-8152072205A8@freeswitch.org>

Definitely! Create a wiki page named Kamailio and add your content  
there. We will then link to it from other appropriate pages.

Thanks!
-MC (IRC:mercutioviz)

Sent from my iPhone

On May 17, 2010, at 2:01 AM, Daniel-Constantin Mierla  
<miconda at gmail.com> wrote:

> Hello,
>
> I put together a tutorial about using kamailio (openser) and  
> freeswtich
> together: the proxy takes care of authentication and registration,
> freeswitch of media services, here is a link:
>
> http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
>
> Is it ok to upload it to FS wiki so others can add to it?
>
> Cheers,
> Daniel
>
> -- 
> Daniel-Constantin Mierla
> Kamailio (OpenSER) Advanced Training
> Miami, Fl, USA - June 21-23, 2010
> http://www.asipto.com/index.php/kamailio-advanced-training/
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From gmaruzz at celliax.org  Mon May 17 09:48:02 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 17 May 2010 18:48:02 +0200
Subject: [Freeswitch-users] kamailio and freeswitch integration
In-Reply-To: <3BA4F163-24A3-4C91-A3B7-8152072205A8@freeswitch.org>
References: <4BF105DA.2060209@gmail.com>
	<3BA4F163-24A3-4C91-A3B7-8152072205A8@freeswitch.org>
Message-ID: <AANLkTikJ1aMGvHv0lgqPdevZB3lCkBhvIzqGT26ldvNE@mail.gmail.com>

Daniel,

your link seems not reachable.
We'll wait for the wiki page :).

Btw, much appreciated, thanks!

-giovanni

On Mon, May 17, 2010 at 4:54 PM, Michael S  Collins <msc at freeswitch.org> wrote:
> Definitely! Create a wiki page named Kamailio and add your content
> there. We will then link to it from other appropriate pages.
>
> Thanks!
> -MC (IRC:mercutioviz)
>
> Sent from my iPhone
>
> On May 17, 2010, at 2:01 AM, Daniel-Constantin Mierla
> <miconda at gmail.com> wrote:
>
>> Hello,
>>
>> I put together a tutorial about using kamailio (openser) and
>> freeswtich
>> together: the proxy takes care of authentication and registration,
>> freeswitch of media services, here is a link:
>>
>> http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
>>
>> Is it ok to upload it to FS wiki so others can add to it?
>>
>> Cheers,
>> Daniel
>>
>> --
>> Daniel-Constantin Mierla
>> Kamailio (OpenSER) Advanced Training
>> Miami, Fl, USA - June 21-23, 2010
>> http://www.asipto.com/index.php/kamailio-advanced-training/
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From pjintheusa at gmail.com  Mon May 17 10:19:03 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Mon, 17 May 2010 13:19:03 -0400
Subject: [Freeswitch-users] kamailio and freeswitch integration
In-Reply-To: <AANLkTikJ1aMGvHv0lgqPdevZB3lCkBhvIzqGT26ldvNE@mail.gmail.com>
References: <4BF105DA.2060209@gmail.com>
	<3BA4F163-24A3-4C91-A3B7-8152072205A8@freeswitch.org>
	<AANLkTikJ1aMGvHv0lgqPdevZB3lCkBhvIzqGT26ldvNE@mail.gmail.com>
Message-ID: <AANLkTin5sPaagA4RglV3JfJ0rTZJy8_FzNwd4IpVx3LJ@mail.gmail.com>

For those that cannot wait (like me!)

http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms

On Mon, May 17, 2010 at 12:48 PM, Giovanni Maruzzelli
<gmaruzz at celliax.org>wrote:

> Daniel,
>
> your link seems not reachable.
> We'll wait for the wiki page :).
>
> Btw, much appreciated, thanks!
>
> -giovanni
>
> On Mon, May 17, 2010 at 4:54 PM, Michael S  Collins <msc at freeswitch.org>
> wrote:
> > Definitely! Create a wiki page named Kamailio and add your content
> > there. We will then link to it from other appropriate pages.
> >
> > Thanks!
> > -MC (IRC:mercutioviz)
> >
> > Sent from my iPhone
> >
> > On May 17, 2010, at 2:01 AM, Daniel-Constantin Mierla
> > <miconda at gmail.com> wrote:
> >
> >> Hello,
> >>
> >> I put together a tutorial about using kamailio (openser) and
> >> freeswtich
> >> together: the proxy takes care of authentication and registration,
> >> freeswitch of media services, here is a link:
> >>
> >> http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
> >>
> >> Is it ok to upload it to FS wiki so others can add to it?
> >>
> >> Cheers,
> >> Daniel
> >>
> >> --
> >> Daniel-Constantin Mierla
> >> Kamailio (OpenSER) Advanced Training
> >> Miami, Fl, USA - June 21-23, 2010
> >> http://www.asipto.com/index.php/kamailio-advanced-training/
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> users
> >> http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From infos at madovsky.org  Mon May 17 10:44:16 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 17 May 2010 13:44:16 -0400
Subject: [Freeswitch-users] kamailio and freeswitch integration
References: <4BF105DA.2060209@gmail.com><3BA4F163-24A3-4C91-A3B7-8152072205A8@freeswitch.org><AANLkTikJ1aMGvHv0lgqPdevZB3lCkBhvIzqGT26ldvNE@mail.gmail.com>
	<AANLkTin5sPaagA4RglV3JfJ0rTZJy8_FzNwd4IpVx3LJ@mail.gmail.com>
Message-ID: <780492B9CF324B359CA987710CA7738D@MOBILEE1705>

doesn't work
  ----- Original Message ----- 
  From: Phillip Jones 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, May 17, 2010 1:19 PM
  Subject: Re: [Freeswitch-users] kamailio and freeswitch integration


  For those that cannot wait (like me!)

  http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms


  On Mon, May 17, 2010 at 12:48 PM, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:

    Daniel,

    your link seems not reachable.
    We'll wait for the wiki page :).

    Btw, much appreciated, thanks!

    -giovanni


    On Mon, May 17, 2010 at 4:54 PM, Michael S  Collins <msc at freeswitch.org> wrote:
    > Definitely! Create a wiki page named Kamailio and add your content
    > there. We will then link to it from other appropriate pages.
    >
    > Thanks!
    > -MC (IRC:mercutioviz)
    >
    > Sent from my iPhone
    >
    > On May 17, 2010, at 2:01 AM, Daniel-Constantin Mierla
    > <miconda at gmail.com> wrote:
    >
    >> Hello,
    >>
    >> I put together a tutorial about using kamailio (openser) and
    >> freeswtich
    >> together: the proxy takes care of authentication and registration,
    >> freeswitch of media services, here is a link:
    >>
    >> http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
    >>
    >> Is it ok to upload it to FS wiki so others can add to it?
    >>
    >> Cheers,
    >> Daniel
    >>
    >> --
    >> Daniel-Constantin Mierla
    >> Kamailio (OpenSER) Advanced Training
    >> Miami, Fl, USA - June 21-23, 2010
    >> http://www.asipto.com/index.php/kamailio-advanced-training/
    >>
    >>
    >> _______________________________________________
    >> FreeSWITCH-users mailing list
    >> FreeSWITCH-users at lists.freeswitch.org
    >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
    >> users
    >> http://www.freeswitch.org
    >
    > _______________________________________________
    > FreeSWITCH-users mailing list
    > FreeSWITCH-users at lists.freeswitch.org
    > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    > http://www.freeswitch.org
    >




    --
    Sincerely,

    Giovanni Maruzzelli
    Cell : +39-347-2665618


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





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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From gmaruzz at celliax.org  Mon May 17 10:48:21 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 17 May 2010 19:48:21 +0200
Subject: [Freeswitch-users] kamailio and freeswitch integration
In-Reply-To: <780492B9CF324B359CA987710CA7738D@MOBILEE1705>
References: <4BF105DA.2060209@gmail.com>
	<3BA4F163-24A3-4C91-A3B7-8152072205A8@freeswitch.org> 
	<AANLkTikJ1aMGvHv0lgqPdevZB3lCkBhvIzqGT26ldvNE@mail.gmail.com> 
	<AANLkTin5sPaagA4RglV3JfJ0rTZJy8_FzNwd4IpVx3LJ@mail.gmail.com> 
	<780492B9CF324B359CA987710CA7738D@MOBILEE1705>
Message-ID: <AANLkTileOmv9Z6pQtQcZuEzPfYHv-I5pkEVpm2OYdVxg@mail.gmail.com>

neither from Serbia

On Mon, May 17, 2010 at 7:44 PM, Madovsky <infos at madovsky.org> wrote:
> doesn't work
>
> ----- Original Message -----
> From: Phillip Jones
> To: freeswitch-users at lists.freeswitch.org
> Sent: Monday, May 17, 2010 1:19 PM
> Subject: Re: [Freeswitch-users] kamailio and freeswitch integration
> For those that cannot wait (like me!)
>
> http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
>
> On Mon, May 17, 2010 at 12:48 PM, Giovanni Maruzzelli <gmaruzz at celliax.org>
> wrote:
>>
>> Daniel,
>>
>> your link seems not reachable.
>> We'll wait for the wiki page :).
>>
>> Btw, much appreciated, thanks!
>>
>> -giovanni
>>
>> On Mon, May 17, 2010 at 4:54 PM, Michael S ?Collins <msc at freeswitch.org>
>> wrote:
>> > Definitely! Create a wiki page named Kamailio and add your content
>> > there. We will then link to it from other appropriate pages.
>> >
>> > Thanks!
>> > -MC (IRC:mercutioviz)
>> >
>> > Sent from my iPhone
>> >
>> > On May 17, 2010, at 2:01 AM, Daniel-Constantin Mierla
>> > <miconda at gmail.com> wrote:
>> >
>> >> Hello,
>> >>
>> >> I put together a tutorial about using kamailio (openser) and
>> >> freeswtich
>> >> together: the proxy takes care of authentication and registration,
>> >> freeswitch of media services, here is a link:
>> >>
>> >> http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
>> >>
>> >> Is it ok to upload it to FS wiki so others can add to it?
>> >>
>> >> Cheers,
>> >> Daniel
>> >>
>> >> --
>> >> Daniel-Constantin Mierla
>> >> Kamailio (OpenSER) Advanced Training
>> >> Miami, Fl, USA - June 21-23, 2010
>> >> http://www.asipto.com/index.php/kamailio-advanced-training/
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> >> users
>> >> http://www.freeswitch.org
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> ________________________________
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From sos at sokhapkin.dyndns.org  Mon May 17 10:55:17 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 17 May 2010 13:55:17 -0400
Subject: [Freeswitch-users] kamailio and freeswitch integration
In-Reply-To: <780492B9CF324B359CA987710CA7738D@MOBILEE1705>
References: <4BF105DA.2060209@gmail.com>
	<AANLkTin5sPaagA4RglV3JfJ0rTZJy8_FzNwd4IpVx3LJ@mail.gmail.com>
	<780492B9CF324B359CA987710CA7738D@MOBILEE1705>
Message-ID: <201005171355.17228.sos@sokhapkin.dyndns.org>

The link works fine to me.

On Monday 17 May 2010, Madovsky wrote:
> doesn't work
>   ----- Original Message -----
>   From: Phillip Jones
>   To: freeswitch-users at lists.freeswitch.org
>   Sent: Monday, May 17, 2010 1:19 PM
>   Subject: Re: [Freeswitch-users] kamailio and freeswitch integration
> 
> 
>   For those that cannot wait (like me!)
> 
>   http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
> 
> 
>   On Mon, May 17, 2010 at 12:48 PM, Giovanni Maruzzelli
>  <gmaruzz at celliax.org> wrote:
> 
>     Daniel,
> 
>     your link seems not reachable.
>     We'll wait for the wiki page :).
> 
>     Btw, much appreciated, thanks!
> 
>     -giovanni
> 
>     On Mon, May 17, 2010 at 4:54 PM, Michael S  Collins <msc at freeswitch.org> 
wrote:
>     > Definitely! Create a wiki page named Kamailio and add your content
>     > there. We will then link to it from other appropriate pages.
>     >
>     > Thanks!
>     > -MC (IRC:mercutioviz)
>     >
>     > Sent from my iPhone
>     >
>     > On May 17, 2010, at 2:01 AM, Daniel-Constantin Mierla
>     >
>     > <miconda at gmail.com> wrote:
>     >> Hello,
>     >>
>     >> I put together a tutorial about using kamailio (openser) and
>     >> freeswtich
>     >> together: the proxy takes care of authentication and registration,
>     >> freeswitch of media services, here is a link:
>     >>
>     >> http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
>     >>
>     >> Is it ok to upload it to FS wiki so others can add to it?
>     >>
>     >> Cheers,
>     >> Daniel
>     >>
>     >> --
>     >> Daniel-Constantin Mierla
>     >> Kamailio (OpenSER) Advanced Training
>     >> Miami, Fl, USA - June 21-23, 2010
>     >> http://www.asipto.com/index.php/kamailio-advanced-training/
>     >>
>     >>
>     >> _______________________________________________
>     >> FreeSWITCH-users mailing list
>     >> FreeSWITCH-users at lists.freeswitch.org
>     >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>     >> users
>     >> http://www.freeswitch.org
>     >
>     > _______________________________________________
>     > FreeSWITCH-users mailing list
>     > FreeSWITCH-users at lists.freeswitch.org
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
>     >ers http://www.freeswitch.org
> 
>     --
>     Sincerely,
> 
>     Giovanni Maruzzelli
>     Cell : +39-347-2665618
> 
> 
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>    
>  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  http://www.freeswitch.org
> 
> 
> 
> 
> 
> ---------------------------------------------------------------------------
> ---
> 
> 
>   _______________________________________________
>   FreeSWITCH-users mailing list
>   FreeSWITCH-users at lists.freeswitch.org
>   http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>   UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>   http://www.freeswitch.org
> 



From infos at madovsky.org  Mon May 17 11:10:57 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 17 May 2010 14:10:57 -0400
Subject: [Freeswitch-users] kamailio and freeswitch integration
References: <4BF105DA.2060209@gmail.com><AANLkTin5sPaagA4RglV3JfJ0rTZJy8_FzNwd4IpVx3LJ@mail.gmail.com><780492B9CF324B359CA987710CA7738D@MOBILEE1705>
	<201005171355.17228.sos@sokhapkin.dyndns.org>
Message-ID: <A3BDB256F37F4E468E8268F529E77081@MOBILEE1705>

weird...
from canada (montreal)
not works

----- Original Message ----- 
From: "Sergey Okhapkin" <sos at sokhapkin.dyndns.org>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Monday, May 17, 2010 1:55 PM
Subject: Re: [Freeswitch-users] kamailio and freeswitch integration


> The link works fine to me.
>
> On Monday 17 May 2010, Madovsky wrote:
>> doesn't work
>>   ----- Original Message -----
>>   From: Phillip Jones
>>   To: freeswitch-users at lists.freeswitch.org
>>   Sent: Monday, May 17, 2010 1:19 PM
>>   Subject: Re: [Freeswitch-users] kamailio and freeswitch integration
>>
>>
>>   For those that cannot wait (like me!)
>>
>>   http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
>>
>>
>>   On Mon, May 17, 2010 at 12:48 PM, Giovanni Maruzzelli
>>  <gmaruzz at celliax.org> wrote:
>>
>>     Daniel,
>>
>>     your link seems not reachable.
>>     We'll wait for the wiki page :).
>>
>>     Btw, much appreciated, thanks!
>>
>>     -giovanni
>>
>>     On Mon, May 17, 2010 at 4:54 PM, Michael S  Collins 
>> <msc at freeswitch.org>
> wrote:
>>     > Definitely! Create a wiki page named Kamailio and add your content
>>     > there. We will then link to it from other appropriate pages.
>>     >
>>     > Thanks!
>>     > -MC (IRC:mercutioviz)
>>     >
>>     > Sent from my iPhone
>>     >
>>     > On May 17, 2010, at 2:01 AM, Daniel-Constantin Mierla
>>     >
>>     > <miconda at gmail.com> wrote:
>>     >> Hello,
>>     >>
>>     >> I put together a tutorial about using kamailio (openser) and
>>     >> freeswtich
>>     >> together: the proxy takes care of authentication and registration,
>>     >> freeswitch of media services, here is a link:
>>     >>
>>     >> 
>> http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
>>     >>
>>     >> Is it ok to upload it to FS wiki so others can add to it?
>>     >>
>>     >> Cheers,
>>     >> Daniel
>>     >>
>>     >> --
>>     >> Daniel-Constantin Mierla
>>     >> Kamailio (OpenSER) Advanced Training
>>     >> Miami, Fl, USA - June 21-23, 2010
>>     >> http://www.asipto.com/index.php/kamailio-advanced-training/
>>     >>
>>     >>
>>     >> _______________________________________________
>>     >> FreeSWITCH-users mailing list
>>     >> FreeSWITCH-users at lists.freeswitch.org
>>     >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     >> 
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>     >> users
>>     >> http://www.freeswitch.org
>>     >
>>     > _______________________________________________
>>     > FreeSWITCH-users mailing list
>>     > FreeSWITCH-users at lists.freeswitch.org
>>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     > 
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
>>     >ers http://www.freeswitch.org
>>
>>     --
>>     Sincerely,
>>
>>     Giovanni Maruzzelli
>>     Cell : +39-347-2665618
>>
>>
>>     _______________________________________________
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>>  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>  http://www.freeswitch.org
>>
>>
>>
>>
>>
>> ---------------------------------------------------------------------------
>> ---
>>
>>
>>   _______________________________________________
>>   FreeSWITCH-users mailing list
>>   FreeSWITCH-users at lists.freeswitch.org
>>   http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> 
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>   http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From pjintheusa at gmail.com  Mon May 17 11:22:57 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Mon, 17 May 2010 14:22:57 -0400
Subject: [Freeswitch-users] kamailio and freeswitch integration
In-Reply-To: <A3BDB256F37F4E468E8268F529E77081@MOBILEE1705>
References: <4BF105DA.2060209@gmail.com>
	<AANLkTin5sPaagA4RglV3JfJ0rTZJy8_FzNwd4IpVx3LJ@mail.gmail.com>
	<780492B9CF324B359CA987710CA7738D@MOBILEE1705>
	<201005171355.17228.sos@sokhapkin.dyndns.org>
	<A3BDB256F37F4E468E8268F529E77081@MOBILEE1705>
Message-ID: <AANLkTikpv0hVHZUgzCR6HsVuR73rmrUnEnOOSXp3pZq6@mail.gmail.com>

I think their server is up and down. Worked for me when I posted. Does not
now. Base URL ok though.

On Mon, May 17, 2010 at 2:10 PM, Madovsky <infos at madovsky.org> wrote:

> weird...
> from canada (montreal)
> not works
>
> ----- Original Message -----
> From: "Sergey Okhapkin" <sos at sokhapkin.dyndns.org>
> To: <freeswitch-users at lists.freeswitch.org>
> Sent: Monday, May 17, 2010 1:55 PM
> Subject: Re: [Freeswitch-users] kamailio and freeswitch integration
>
>
> > The link works fine to me.
> >
> > On Monday 17 May 2010, Madovsky wrote:
> >> doesn't work
> >>   ----- Original Message -----
> >>   From: Phillip Jones
> >>   To: freeswitch-users at lists.freeswitch.org
> >>   Sent: Monday, May 17, 2010 1:19 PM
> >>   Subject: Re: [Freeswitch-users] kamailio and freeswitch integration
> >>
> >>
> >>   For those that cannot wait (like me!)
> >>
> >>   http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
> >>
> >>
> >>   On Mon, May 17, 2010 at 12:48 PM, Giovanni Maruzzelli
> >>  <gmaruzz at celliax.org> wrote:
> >>
> >>     Daniel,
> >>
> >>     your link seems not reachable.
> >>     We'll wait for the wiki page :).
> >>
> >>     Btw, much appreciated, thanks!
> >>
> >>     -giovanni
> >>
> >>     On Mon, May 17, 2010 at 4:54 PM, Michael S  Collins
> >> <msc at freeswitch.org>
> > wrote:
> >>     > Definitely! Create a wiki page named Kamailio and add your content
> >>     > there. We will then link to it from other appropriate pages.
> >>     >
> >>     > Thanks!
> >>     > -MC (IRC:mercutioviz)
> >>     >
> >>     > Sent from my iPhone
> >>     >
> >>     > On May 17, 2010, at 2:01 AM, Daniel-Constantin Mierla
> >>     >
> >>     > <miconda at gmail.com> wrote:
> >>     >> Hello,
> >>     >>
> >>     >> I put together a tutorial about using kamailio (openser) and
> >>     >> freeswtich
> >>     >> together: the proxy takes care of authentication and
> registration,
> >>     >> freeswitch of media services, here is a link:
> >>     >>
> >>     >>
> >> http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
> >>     >>
> >>     >> Is it ok to upload it to FS wiki so others can add to it?
> >>     >>
> >>     >> Cheers,
> >>     >> Daniel
> >>     >>
> >>     >> --
> >>     >> Daniel-Constantin Mierla
> >>     >> Kamailio (OpenSER) Advanced Training
> >>     >> Miami, Fl, USA - June 21-23, 2010
> >>     >> http://www.asipto.com/index.php/kamailio-advanced-training/
> >>     >>
> >>     >>
> >>     >> _______________________________________________
> >>     >> FreeSWITCH-users mailing list
> >>     >> FreeSWITCH-users at lists.freeswitch.org
> >>     >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>     >>
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >>     >> users
> >>     >> http://www.freeswitch.org
> >>     >
> >>     > _______________________________________________
> >>     > FreeSWITCH-users mailing list
> >>     > FreeSWITCH-users at lists.freeswitch.org
> >>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>     >
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
> >>     >ers http://www.freeswitch.org
> >>
> >>     --
> >>     Sincerely,
> >>
> >>     Giovanni Maruzzelli
> >>     Cell : +39-347-2665618
> >>
> >>
> >>     _______________________________________________
> >>     FreeSWITCH-users mailing list
> >>     FreeSWITCH-users at lists.freeswitch.org
> >>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>
> >>  UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>  http://www.freeswitch.org
> >>
> >>
> >>
> >>
> >>
> >>
> ---------------------------------------------------------------------------
> >> ---
> >>
> >>
> >>   _______________________________________________
> >>   FreeSWITCH-users mailing list
> >>   FreeSWITCH-users at lists.freeswitch.org
> >>   http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>   http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From pjintheusa at gmail.com  Mon May 17 11:27:57 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Mon, 17 May 2010 14:27:57 -0400
Subject: [Freeswitch-users] Using mod_managed for directory info / DRK
	presentation
Message-ID: <AANLkTikvnO0lCvNi8wc8r_DCw18BurTJwHEpaa1y5V6o@mail.gmail.com>

In DRK presentations last conference call, he showed how to hook into the
registered user / directory lookup by setting a callback from with in
mod_manged code.

Can anyone you saw the presentation (or otherwise) tell me how to do this.
It looked so easy at the time but now I am stumped.

Thanks

Phil
-------------- next part --------------
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From talk2ram at gmail.com  Mon May 17 11:43:09 2010
From: talk2ram at gmail.com (ram)
Date: Tue, 18 May 2010 00:13:09 +0530
Subject: [Freeswitch-users] kamailio and freeswitch integration
In-Reply-To: <AANLkTikpv0hVHZUgzCR6HsVuR73rmrUnEnOOSXp3pZq6@mail.gmail.com>
References: <4BF105DA.2060209@gmail.com>
	<AANLkTin5sPaagA4RglV3JfJ0rTZJy8_FzNwd4IpVx3LJ@mail.gmail.com>
	<780492B9CF324B359CA987710CA7738D@MOBILEE1705>
	<201005171355.17228.sos@sokhapkin.dyndns.org>
	<A3BDB256F37F4E468E8268F529E77081@MOBILEE1705>
	<AANLkTikpv0hVHZUgzCR6HsVuR73rmrUnEnOOSXp3pZq6@mail.gmail.com>
Message-ID: <AANLkTiktXynfJAtzc-Ks096_U7sfCiX2Z2KbA8fJVVw5@mail.gmail.com>

its down

On Mon, May 17, 2010 at 11:52 PM, Phillip Jones <pjintheusa at gmail.com>wrote:

> I think their server is up and down. Worked for me when I posted. Does not
> now. Base URL ok though.
>
> On Mon, May 17, 2010 at 2:10 PM, Madovsky <infos at madovsky.org> wrote:
>
>> weird...
>> from canada (montreal)
>> not works
>>
>> ----- Original Message -----
>> From: "Sergey Okhapkin" <sos at sokhapkin.dyndns.org>
>> To: <freeswitch-users at lists.freeswitch.org>
>>  Sent: Monday, May 17, 2010 1:55 PM
>> Subject: Re: [Freeswitch-users] kamailio and freeswitch integration
>>
>>
>> > The link works fine to me.
>> >
>> > On Monday 17 May 2010, Madovsky wrote:
>> >> doesn't work
>> >>   ----- Original Message -----
>> >>   From: Phillip Jones
>> >>   To: freeswitch-users at lists.freeswitch.org
>> >>   Sent: Monday, May 17, 2010 1:19 PM
>> >>   Subject: Re: [Freeswitch-users] kamailio and freeswitch integration
>> >>
>> >>
>> >>   For those that cannot wait (like me!)
>> >>
>> >>   http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
>> >>
>> >>
>> >>   On Mon, May 17, 2010 at 12:48 PM, Giovanni Maruzzelli
>> >>  <gmaruzz at celliax.org> wrote:
>> >>
>> >>     Daniel,
>> >>
>> >>     your link seems not reachable.
>> >>     We'll wait for the wiki page :).
>> >>
>> >>     Btw, much appreciated, thanks!
>> >>
>> >>     -giovanni
>> >>
>> >>     On Mon, May 17, 2010 at 4:54 PM, Michael S  Collins
>> >> <msc at freeswitch.org>
>> > wrote:
>> >>     > Definitely! Create a wiki page named Kamailio and add your
>> content
>> >>     > there. We will then link to it from other appropriate pages.
>> >>     >
>> >>     > Thanks!
>> >>     > -MC (IRC:mercutioviz)
>> >>     >
>> >>     > Sent from my iPhone
>> >>     >
>> >>     > On May 17, 2010, at 2:01 AM, Daniel-Constantin Mierla
>> >>     >
>> >>     > <miconda at gmail.com> wrote:
>> >>     >> Hello,
>> >>     >>
>> >>     >> I put together a tutorial about using kamailio (openser) and
>> >>     >> freeswtich
>> >>     >> together: the proxy takes care of authentication and
>> registration,
>> >>     >> freeswitch of media services, here is a link:
>> >>     >>
>> >>     >>
>> >> http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
>> >>     >>
>> >>     >> Is it ok to upload it to FS wiki so others can add to it?
>> >>     >>
>> >>     >> Cheers,
>> >>     >> Daniel
>> >>     >>
>> >>     >> --
>> >>     >> Daniel-Constantin Mierla
>> >>     >> Kamailio (OpenSER) Advanced Training
>> >>     >> Miami, Fl, USA - June 21-23, 2010
>> >>     >> http://www.asipto.com/index.php/kamailio-advanced-training/
>> >>     >>
>> >>     >>
>> >>     >> _______________________________________________
>> >>     >> FreeSWITCH-users mailing list
>> >>     >> FreeSWITCH-users at lists.freeswitch.org
>> >>     >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>     >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> >>     >> users
>> >>     >> http://www.freeswitch.org
>> >>     >
>> >>     > _______________________________________________
>> >>     > FreeSWITCH-users mailing list
>> >>     > FreeSWITCH-users at lists.freeswitch.org
>> >>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>     >
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
>> >>     >ers http://www.freeswitch.org
>> >>
>> >>     --
>> >>     Sincerely,
>> >>
>> >>     Giovanni Maruzzelli
>> >>     Cell : +39-347-2665618
>> >>
>> >>
>> >>     _______________________________________________
>> >>     FreeSWITCH-users mailing list
>> >>     FreeSWITCH-users at lists.freeswitch.org
>> >>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >>  UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>  http://www.freeswitch.org
>> >>
>> >>
>> >>
>> >>
>> >>
>> >>
>> ---------------------------------------------------------------------------
>> >> ---
>> >>
>> >>
>> >>   _______________________________________________
>> >>   FreeSWITCH-users mailing list
>> >>   FreeSWITCH-users at lists.freeswitch.org
>> >>   http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>   http://www.freeswitch.org
>> >>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Mon May 17 11:49:45 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 17 May 2010 14:49:45 -0400
Subject: [Freeswitch-users] kamailio and freeswitch integration
References: <4BF105DA.2060209@gmail.com><AANLkTin5sPaagA4RglV3JfJ0rTZJy8_FzNwd4IpVx3LJ@mail.gmail.com><780492B9CF324B359CA987710CA7738D@MOBILEE1705><201005171355.17228.sos@sokhapkin.dyndns.org><A3BDB256F37F4E468E8268F529E77081@MOBILEE1705>
	<AANLkTikpv0hVHZUgzCR6HsVuR73rmrUnEnOOSXp3pZq6@mail.gmail.com>
Message-ID: <500DFBB8420E46DEAE5A58BAFDA5066C@MOBILEE1705>

I think the problem is the " : "  in the link....
  ----- Original Message ----- 
  From: Phillip Jones 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, May 17, 2010 2:22 PM
  Subject: Re: [Freeswitch-users] kamailio and freeswitch integration


  I think their server is up and down. Worked for me when I posted. Does not now. Base URL ok though. 


  On Mon, May 17, 2010 at 2:10 PM, Madovsky <infos at madovsky.org> wrote:

    weird...
    from canada (montreal)
    not works


    ----- Original Message -----
    From: "Sergey Okhapkin" <sos at sokhapkin.dyndns.org>
    To: <freeswitch-users at lists.freeswitch.org>

    Sent: Monday, May 17, 2010 1:55 PM
    Subject: Re: [Freeswitch-users] kamailio and freeswitch integration


    > The link works fine to me.
    >
    > On Monday 17 May 2010, Madovsky wrote:
    >> doesn't work
    >>   ----- Original Message -----
    >>   From: Phillip Jones
    >>   To: freeswitch-users at lists.freeswitch.org
    >>   Sent: Monday, May 17, 2010 1:19 PM
    >>   Subject: Re: [Freeswitch-users] kamailio and freeswitch integration
    >>
    >>
    >>   For those that cannot wait (like me!)
    >>
    >>   http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
    >>
    >>
    >>   On Mon, May 17, 2010 at 12:48 PM, Giovanni Maruzzelli
    >>  <gmaruzz at celliax.org> wrote:
    >>
    >>     Daniel,
    >>
    >>     your link seems not reachable.
    >>     We'll wait for the wiki page :).
    >>
    >>     Btw, much appreciated, thanks!
    >>
    >>     -giovanni
    >>
    >>     On Mon, May 17, 2010 at 4:54 PM, Michael S  Collins
    >> <msc at freeswitch.org>
    > wrote:
    >>     > Definitely! Create a wiki page named Kamailio and add your content
    >>     > there. We will then link to it from other appropriate pages.
    >>     >
    >>     > Thanks!
    >>     > -MC (IRC:mercutioviz)
    >>     >
    >>     > Sent from my iPhone
    >>     >
    >>     > On May 17, 2010, at 2:01 AM, Daniel-Constantin Mierla
    >>     >
    >>     > <miconda at gmail.com> wrote:
    >>     >> Hello,
    >>     >>
    >>     >> I put together a tutorial about using kamailio (openser) and
    >>     >> freeswtich
    >>     >> together: the proxy takes care of authentication and registration,
    >>     >> freeswitch of media services, here is a link:
    >>     >>
    >>     >>
    >> http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
    >>     >>
    >>     >> Is it ok to upload it to FS wiki so others can add to it?
    >>     >>
    >>     >> Cheers,
    >>     >> Daniel
    >>     >>
    >>     >> --
    >>     >> Daniel-Constantin Mierla
    >>     >> Kamailio (OpenSER) Advanced Training
    >>     >> Miami, Fl, USA - June 21-23, 2010
    >>     >> http://www.asipto.com/index.php/kamailio-advanced-training/
    >>     >>
    >>     >>
    >>     >> _______________________________________________
    >>     >> FreeSWITCH-users mailing list
    >>     >> FreeSWITCH-users at lists.freeswitch.org
    >>     >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    >>     >>
    >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
    >>     >> users
    >>     >> http://www.freeswitch.org
    >>     >
    >>     > _______________________________________________
    >>     > FreeSWITCH-users mailing list
    >>     > FreeSWITCH-users at lists.freeswitch.org
    >>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    >>     >
    >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
    >>     >ers http://www.freeswitch.org
    >>
    >>     --
    >>     Sincerely,
    >>
    >>     Giovanni Maruzzelli
    >>     Cell : +39-347-2665618
    >>
    >>
    >>     _______________________________________________
    >>     FreeSWITCH-users mailing list
    >>     FreeSWITCH-users at lists.freeswitch.org
    >>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    >>
    >>  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    >>  http://www.freeswitch.org
    >>
    >>
    >>
    >>
    >>
    >> ---------------------------------------------------------------------------
    >> ---
    >>
    >>
    >>   _______________________________________________
    >>   FreeSWITCH-users mailing list
    >>   FreeSWITCH-users at lists.freeswitch.org
    >>   http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    >>
    >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    >>   http://www.freeswitch.org
    >>
    >
    >
    > _______________________________________________
    > FreeSWITCH-users mailing list
    > FreeSWITCH-users at lists.freeswitch.org
    > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
    > http://www.freeswitch.org


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





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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From pjintheusa at gmail.com  Mon May 17 11:52:03 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Mon, 17 May 2010 14:52:03 -0400
Subject: [Freeswitch-users] Using mod_managed for directory info / DRK
	presentation
In-Reply-To: <AANLkTikvnO0lCvNi8wc8r_DCw18BurTJwHEpaa1y5V6o@mail.gmail.com>
References: <AANLkTikvnO0lCvNi8wc8r_DCw18BurTJwHEpaa1y5V6o@mail.gmail.com>
Message-ID: <AANLkTimb6KlWvXoaJ89vSj4sFUw0yxQESS4Czu6eocAH@mail.gmail.com>

Scrub this. I have the info I need.

Thanks.

On Mon, May 17, 2010 at 2:27 PM, Phillip Jones <pjintheusa at gmail.com> wrote:

> In DRK presentations last conference call, he showed how to hook into the
> registered user / directory lookup by setting a callback from with in
> mod_manged code.
>
> Can anyone you saw the presentation (or otherwise) tell me how to do this.
> It looked so easy at the time but now I am stumped.
>
> Thanks
>
> Phil
>
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From talk2ram at gmail.com  Mon May 17 12:03:49 2010
From: talk2ram at gmail.com (ram)
Date: Tue, 18 May 2010 00:33:49 +0530
Subject: [Freeswitch-users] kamailio and freeswitch integration
In-Reply-To: <500DFBB8420E46DEAE5A58BAFDA5066C@MOBILEE1705>
References: <4BF105DA.2060209@gmail.com>
	<AANLkTin5sPaagA4RglV3JfJ0rTZJy8_FzNwd4IpVx3LJ@mail.gmail.com>
	<780492B9CF324B359CA987710CA7738D@MOBILEE1705>
	<201005171355.17228.sos@sokhapkin.dyndns.org>
	<A3BDB256F37F4E468E8268F529E77081@MOBILEE1705>
	<AANLkTikpv0hVHZUgzCR6HsVuR73rmrUnEnOOSXp3pZq6@mail.gmail.com>
	<500DFBB8420E46DEAE5A58BAFDA5066C@MOBILEE1705>
Message-ID: <AANLkTikc28zn24mmkprrlrUbwyDKshxY0AEPvQo2uFrj@mail.gmail.com>

http://kb.asipto.com/freeswitchkamailio-3.0.x-freeswitch-1.0.6d-ms

is this one correct ?

Ram

On Tue, May 18, 2010 at 12:19 AM, Madovsky <infos at madovsky.org> wrote:

>  I think the problem is the " : "  in the link....
>
>  ----- Original Message -----
> *From:* Phillip Jones <pjintheusa at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
>   *Sent:* Monday, May 17, 2010 2:22 PM
> *Subject:* Re: [Freeswitch-users] kamailio and freeswitch integration
>
> I think their server is up and down. Worked for me when I posted. Does not
> now. Base URL ok though.
>
> On Mon, May 17, 2010 at 2:10 PM, Madovsky <infos at madovsky.org> wrote:
>
>> weird...
>> from canada (montreal)
>> not works
>>
>> ----- Original Message -----
>> From: "Sergey Okhapkin" <sos at sokhapkin.dyndns.org>
>> To: <freeswitch-users at lists.freeswitch.org>
>>  Sent: Monday, May 17, 2010 1:55 PM
>> Subject: Re: [Freeswitch-users] kamailio and freeswitch integration
>>
>>
>> > The link works fine to me.
>> >
>> > On Monday 17 May 2010, Madovsky wrote:
>> >> doesn't work
>> >>   ----- Original Message -----
>> >>   From: Phillip Jones
>> >>   To: freeswitch-users at lists.freeswitch.org
>> >>   Sent: Monday, May 17, 2010 1:19 PM
>> >>   Subject: Re: [Freeswitch-users] kamailio and freeswitch integration
>> >>
>> >>
>> >>   For those that cannot wait (like me!)
>> >>
>> >>   http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
>> >>
>> >>
>> >>   On Mon, May 17, 2010 at 12:48 PM, Giovanni Maruzzelli
>> >>  <gmaruzz at celliax.org> wrote:
>> >>
>> >>     Daniel,
>> >>
>> >>     your link seems not reachable.
>> >>     We'll wait for the wiki page :).
>> >>
>> >>     Btw, much appreciated, thanks!
>> >>
>> >>     -giovanni
>> >>
>> >>     On Mon, May 17, 2010 at 4:54 PM, Michael S  Collins
>> >> <msc at freeswitch.org>
>> > wrote:
>> >>     > Definitely! Create a wiki page named Kamailio and add your
>> content
>> >>     > there. We will then link to it from other appropriate pages.
>> >>     >
>> >>     > Thanks!
>> >>     > -MC (IRC:mercutioviz)
>> >>     >
>> >>     > Sent from my iPhone
>> >>     >
>> >>     > On May 17, 2010, at 2:01 AM, Daniel-Constantin Mierla
>> >>     >
>> >>     > <miconda at gmail.com> wrote:
>> >>     >> Hello,
>> >>     >>
>> >>     >> I put together a tutorial about using kamailio (openser) and
>> >>     >> freeswtich
>> >>     >> together: the proxy takes care of authentication and
>> registration,
>> >>     >> freeswitch of media services, here is a link:
>> >>     >>
>> >>     >>
>> >> http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
>> >>     >>
>> >>     >> Is it ok to upload it to FS wiki so others can add to it?
>> >>     >>
>> >>     >> Cheers,
>> >>     >> Daniel
>> >>     >>
>> >>     >> --
>> >>     >> Daniel-Constantin Mierla
>> >>     >> Kamailio (OpenSER) Advanced Training
>> >>     >> Miami, Fl, USA - June 21-23, 2010
>> >>     >> http://www.asipto.com/index.php/kamailio-advanced-training/
>> >>     >>
>> >>     >>
>> >>     >> _______________________________________________
>> >>     >> FreeSWITCH-users mailing list
>> >>     >> FreeSWITCH-users at lists.freeswitch.org
>> >>     >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>     >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> >>     >> users
>> >>     >> http://www.freeswitch.org
>> >>     >
>> >>     > _______________________________________________
>> >>     > FreeSWITCH-users mailing list
>> >>     > FreeSWITCH-users at lists.freeswitch.org
>> >>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>     >
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
>> >>     >ers http://www.freeswitch.org
>> >>
>> >>     --
>> >>     Sincerely,
>> >>
>> >>     Giovanni Maruzzelli
>> >>     Cell : +39-347-2665618
>> >>
>> >>
>> >>     _______________________________________________
>> >>     FreeSWITCH-users mailing list
>> >>     FreeSWITCH-users at lists.freeswitch.org
>> >>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >>  UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>  http://www.freeswitch.org
>> >>
>> >>
>> >>
>> >>
>> >>
>> >>
>> ---------------------------------------------------------------------------
>> >> ---
>> >>
>> >>
>> >>   _______________________________________________
>> >>   FreeSWITCH-users mailing list
>> >>   FreeSWITCH-users at lists.freeswitch.org
>> >>   http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>   http://www.freeswitch.org
>> >>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From miconda at gmail.com  Mon May 17 12:04:36 2010
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Mon, 17 May 2010 21:04:36 +0200
Subject: [Freeswitch-users] kamailio and freeswitch integration
In-Reply-To: <500DFBB8420E46DEAE5A58BAFDA5066C@MOBILEE1705>
References: <4BF105DA.2060209@gmail.com><AANLkTin5sPaagA4RglV3JfJ0rTZJy8_FzNwd4IpVx3LJ@mail.gmail.com><780492B9CF324B359CA987710CA7738D@MOBILEE1705><201005171355.17228.sos@sokhapkin.dyndns.org><A3BDB256F37F4E468E8268F529E77081@MOBILEE1705>	<AANLkTikpv0hVHZUgzCR6HsVuR73rmrUnEnOOSXp3pZq6@mail.gmail.com>
	<500DFBB8420E46DEAE5A58BAFDA5066C@MOBILEE1705>
Message-ID: <4BF19344.4090607@gmail.com>

Hello,

sorry for inconveniences, but dns propagation really sucks big this time 
-- record is more than one week old and seems some parts of the world 
still have an older cache. Being in travel to sipit26 cannot invest much 
time in it and copy to fs wiki, so i just redirected one of older 
records to same site, use this link for a while:

http://ngs.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms

Colon - ':' -  in link is not a problem.

@Phillip Jones - probably you got exactly some restart of the web server 
while redirecting the subdomain, site is up 99.999%.

Cheers,
Daniel


On 5/17/10 8:49 PM, Madovsky wrote:
> I think the problem is the " : "  in the link....
>
>     ----- Original Message -----
>     *From:* Phillip Jones <mailto:pjintheusa at gmail.com>
>     *To:* freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>
>     *Sent:* Monday, May 17, 2010 2:22 PM
>     *Subject:* Re: [Freeswitch-users] kamailio and freeswitch integration
>
>     I think their server is up and down. Worked for me when I posted.
>     Does not now. Base URL ok though.
>
>     On Mon, May 17, 2010 at 2:10 PM, Madovsky <infos at madovsky.org
>     <mailto:infos at madovsky.org>> wrote:
>
>         weird...
>         from canada (montreal)
>         not works
>
>         ----- Original Message -----
>         From: "Sergey Okhapkin" <sos at sokhapkin.dyndns.org
>         <mailto:sos at sokhapkin.dyndns.org>>
>         To: <freeswitch-users at lists.freeswitch.org
>         <mailto:freeswitch-users at lists.freeswitch.org>>
>         Sent: Monday, May 17, 2010 1:55 PM
>         Subject: Re: [Freeswitch-users] kamailio and freeswitch
>         integration
>
>
>         > The link works fine to me.
>         >
>         > On Monday 17 May 2010, Madovsky wrote:
>         >> doesn't work
>         >>   ----- Original Message -----
>         >>   From: Phillip Jones
>         >>   To: freeswitch-users at lists.freeswitch.org
>         <mailto:freeswitch-users at lists.freeswitch.org>
>         >>   Sent: Monday, May 17, 2010 1:19 PM
>         >>   Subject: Re: [Freeswitch-users] kamailio and freeswitch
>         integration
>         >>
>         >>
>         >>   For those that cannot wait (like me!)
>         >>
>         >>
>         http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
>         >>
>         >>
>         >>   On Mon, May 17, 2010 at 12:48 PM, Giovanni Maruzzelli
>         >> <gmaruzz at celliax.org <mailto:gmaruzz at celliax.org>> wrote:
>         >>
>         >>     Daniel,
>         >>
>         >>     your link seems not reachable.
>         >>     We'll wait for the wiki page :).
>         >>
>         >>     Btw, much appreciated, thanks!
>         >>
>         >>     -giovanni
>         >>
>         >>     On Mon, May 17, 2010 at 4:54 PM, Michael S  Collins
>         >> <msc at freeswitch.org <mailto:msc at freeswitch.org>>
>         > wrote:
>         >> > Definitely! Create a wiki page named Kamailio and add
>         your content
>         >> > there. We will then link to it from other appropriate pages.
>         >> >
>         >> > Thanks!
>         >> > -MC (IRC:mercutioviz)
>         >> >
>         >> > Sent from my iPhone
>         >> >
>         >> > On May 17, 2010, at 2:01 AM, Daniel-Constantin Mierla
>         >> >
>         >> > <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>         >> >> Hello,
>         >> >>
>         >> >> I put together a tutorial about using kamailio (openser) and
>         >> >> freeswtich
>         >> >> together: the proxy takes care of authentication and
>         registration,
>         >> >> freeswitch of media services, here is a link:
>         >> >>
>         >> >>
>         >>
>         http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
>         >> >>
>         >> >> Is it ok to upload it to FS wiki so others can add to it?
>         >> >>
>         >> >> Cheers,
>         >> >> Daniel
>         >> >>
>         >> >> --
>         >> >> Daniel-Constantin Mierla
>         >> >> Kamailio (OpenSER) Advanced Training
>         >> >> Miami, Fl, USA - June 21-23, 2010
>         >> >> http://www.asipto.com/index.php/kamailio-advanced-training/
>         >> >>
>         >> >>
>         >> >> _______________________________________________
>         >> >> FreeSWITCH-users mailing list
>         >> >> FreeSWITCH-users at lists.freeswitch.org
>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>         >> >>
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         >> >>
>         >>
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>         >> >> users
>         >> >> http://www.freeswitch.org
>         >> >
>         >> > _______________________________________________
>         >> > FreeSWITCH-users mailing list
>         >> > FreeSWITCH-users at lists.freeswitch.org
>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>         >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         >> >
>         >>
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
>         >> >ers http://www.freeswitch.org
>         >>
>         >>     --
>         >>     Sincerely,
>         >>
>         >>     Giovanni Maruzzelli
>         >>     Cell : +39-347-2665618
>         >>
>         >>
>         >>     _______________________________________________
>         >>     FreeSWITCH-users mailing list
>         >> FreeSWITCH-users at lists.freeswitch.org
>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>         >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         >>
>         >>
>          UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         >> http://www.freeswitch.org
>         >>
>         >>
>         >>
>         >>
>         >>
>         >>
>         ---------------------------------------------------------------------------
>         >> ---
>         >>
>         >>
>         >>   _______________________________________________
>         >>   FreeSWITCH-users mailing list
>         >> FreeSWITCH-users at lists.freeswitch.org
>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>         >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         >>
>         >>
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         >> http://www.freeswitch.org
>         >>
>         >
>         >
>         > _______________________________________________
>         > FreeSWITCH-users mailing list
>         > FreeSWITCH-users at lists.freeswitch.org
>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>         > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         >
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         > http://www.freeswitch.org
>
>
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>
>
>     ------------------------------------------------------------------------
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    

-- 
Daniel-Constantin Mierla
Kamailio (OpenSER) Advanced Training
Miami, Fl, USA - June 21-23, 2010
http://www.asipto.com/index.php/kamailio-advanced-training/

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From david.ponzone at gmail.com  Mon May 17 12:04:36 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Mon, 17 May 2010 21:04:36 +0200
Subject: [Freeswitch-users] kamailio and freeswitch integration
In-Reply-To: <500DFBB8420E46DEAE5A58BAFDA5066C@MOBILEE1705>
References: <4BF105DA.2060209@gmail.com><AANLkTin5sPaagA4RglV3JfJ0rTZJy8_FzNwd4IpVx3LJ@mail.gmail.com><780492B9CF324B359CA987710CA7738D@MOBILEE1705><201005171355.17228.sos@sokhapkin.dyndns.org><A3BDB256F37F4E468E8268F529E77081@MOBILEE1705>
	<AANLkTikpv0hVHZUgzCR6HsVuR73rmrUnEnOOSXp3pZq6@mail.gmail.com>
	<500DFBB8420E46DEAE5A58BAFDA5066C@MOBILEE1705>
Message-ID: <033DCE8D-972A-4507-AC66-5D3028CA882A@gmail.com>

Works fine here, with Safari, Chrome and Firefox.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 17/05/2010 ? 20:49, Madovsky a ?crit :

> I think the problem is the " : "  in the link....
> ----- Original Message -----
> From: Phillip Jones
> To: freeswitch-users at lists.freeswitch.org
> Sent: Monday, May 17, 2010 2:22 PM
> Subject: Re: [Freeswitch-users] kamailio and freeswitch integration
>
> I think their server is up and down. Worked for me when I posted.  
> Does not now. Base URL ok though.
>
> On Mon, May 17, 2010 at 2:10 PM, Madovsky <infos at madovsky.org> wrote:
> weird...
> from canada (montreal)
> not works
>
> ----- Original Message -----
> From: "Sergey Okhapkin" <sos at sokhapkin.dyndns.org>
> To: <freeswitch-users at lists.freeswitch.org>
> Sent: Monday, May 17, 2010 1:55 PM
> Subject: Re: [Freeswitch-users] kamailio and freeswitch integration
>
>
> > The link works fine to me.
> >
> > On Monday 17 May 2010, Madovsky wrote:
> >> doesn't work
> >>   ----- Original Message -----
> >>   From: Phillip Jones
> >>   To: freeswitch-users at lists.freeswitch.org
> >>   Sent: Monday, May 17, 2010 1:19 PM
> >>   Subject: Re: [Freeswitch-users] kamailio and freeswitch  
> integration
> >>
> >>
> >>   For those that cannot wait (like me!)
> >>
> >>   http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
> >>
> >>
> >>   On Mon, May 17, 2010 at 12:48 PM, Giovanni Maruzzelli
> >>  <gmaruzz at celliax.org> wrote:
> >>
> >>     Daniel,
> >>
> >>     your link seems not reachable.
> >>     We'll wait for the wiki page :).
> >>
> >>     Btw, much appreciated, thanks!
> >>
> >>     -giovanni
> >>
> >>     On Mon, May 17, 2010 at 4:54 PM, Michael S  Collins
> >> <msc at freeswitch.org>
> > wrote:
> >>     > Definitely! Create a wiki page named Kamailio and add your  
> content
> >>     > there. We will then link to it from other appropriate pages.
> >>     >
> >>     > Thanks!
> >>     > -MC (IRC:mercutioviz)
> >>     >
> >>     > Sent from my iPhone
> >>     >
> >>     > On May 17, 2010, at 2:01 AM, Daniel-Constantin Mierla
> >>     >
> >>     > <miconda at gmail.com> wrote:
> >>     >> Hello,
> >>     >>
> >>     >> I put together a tutorial about using kamailio (openser) and
> >>     >> freeswtich
> >>     >> together: the proxy takes care of authentication and  
> registration,
> >>     >> freeswitch of media services, here is a link:
> >>     >>
> >>     >>
> >> http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
> >>     >>
> >>     >> Is it ok to upload it to FS wiki so others can add to it?
> >>     >>
> >>     >> Cheers,
> >>     >> Daniel
> >>     >>
> >>     >> --
> >>     >> Daniel-Constantin Mierla
> >>     >> Kamailio (OpenSER) Advanced Training
> >>     >> Miami, Fl, USA - June 21-23, 2010
> >>     >> http://www.asipto.com/index.php/kamailio-advanced-training/
> >>     >>
> >>     >>
> >>     >> _______________________________________________
> >>     >> FreeSWITCH-users mailing list
> >>     >> FreeSWITCH-users at lists.freeswitch.org
> >>     >> http://lists.freeswitch.org/mailman/listinfo/freeswitch- 
> users
> >>     >>
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >>     >> users
> >>     >> http://www.freeswitch.org
> >>     >
> >>     > _______________________________________________
> >>     > FreeSWITCH-users mailing list
> >>     > FreeSWITCH-users at lists.freeswitch.org
> >>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>     >
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
> >>     >ers http://www.freeswitch.org
> >>
> >>     --
> >>     Sincerely,
> >>
> >>     Giovanni Maruzzelli
> >>     Cell : +39-347-2665618
> >>
> >>
> >>     _______________________________________________
> >>     FreeSWITCH-users mailing list
> >>     FreeSWITCH-users at lists.freeswitch.org
> >>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>
> >>  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>  http://www.freeswitch.org
> >>
> >>
> >>
> >>
> >>
> >>  
> ---------------------------------------------------------------------------
> >> ---
> >>
> >>
> >>   _______________________________________________
> >>   FreeSWITCH-users mailing list
> >>   FreeSWITCH-users at lists.freeswitch.org
> >>   http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>   http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From infos at madovsky.org  Mon May 17 12:14:47 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 17 May 2010 15:14:47 -0400
Subject: [Freeswitch-users] kamailio and freeswitch integration
References: <4BF105DA.2060209@gmail.com><AANLkTin5sPaagA4RglV3JfJ0rTZJy8_FzNwd4IpVx3LJ@mail.gmail.com><780492B9CF324B359CA987710CA7738D@MOBILEE1705><201005171355.17228.sos@sokhapkin.dyndns.org><A3BDB256F37F4E468E8268F529E77081@MOBILEE1705><AANLkTikpv0hVHZUgzCR6HsVuR73rmrUnEnOOSXp3pZq6@mail.gmail.com><500DFBB8420E46DEAE5A58BAFDA5066C@MOBILEE1705>
	<AANLkTikc28zn24mmkprrlrUbwyDKshxY0AEPvQo2uFrj@mail.gmail.com>
Message-ID: <5620E34753674B9DACDC4898A9A2EE4F@MOBILEE1705>

neither :(
  ----- Original Message ----- 
  From: ram 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, May 17, 2010 3:03 PM
  Subject: Re: [Freeswitch-users] kamailio and freeswitch integration


  http://kb.asipto.com/freeswitchkamailio-3.0.x-freeswitch-1.0.6d-ms

  is this one correct ?

  Ram


  On Tue, May 18, 2010 at 12:19 AM, Madovsky <infos at madovsky.org> wrote:

    I think the problem is the " : "  in the link....
      ----- Original Message ----- 
      From: Phillip Jones 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Monday, May 17, 2010 2:22 PM
      Subject: Re: [Freeswitch-users] kamailio and freeswitch integration


      I think their server is up and down. Worked for me when I posted. Does not now. Base URL ok though. 


      On Mon, May 17, 2010 at 2:10 PM, Madovsky <infos at madovsky.org> wrote:

        weird...
        from canada (montreal)
        not works


        ----- Original Message -----
        From: "Sergey Okhapkin" <sos at sokhapkin.dyndns.org>
        To: <freeswitch-users at lists.freeswitch.org>

        Sent: Monday, May 17, 2010 1:55 PM
        Subject: Re: [Freeswitch-users] kamailio and freeswitch integration


        > The link works fine to me.
        >
        > On Monday 17 May 2010, Madovsky wrote:
        >> doesn't work
        >>   ----- Original Message -----
        >>   From: Phillip Jones
        >>   To: freeswitch-users at lists.freeswitch.org
        >>   Sent: Monday, May 17, 2010 1:19 PM
        >>   Subject: Re: [Freeswitch-users] kamailio and freeswitch integration
        >>
        >>
        >>   For those that cannot wait (like me!)
        >>
        >>   http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
        >>
        >>
        >>   On Mon, May 17, 2010 at 12:48 PM, Giovanni Maruzzelli
        >>  <gmaruzz at celliax.org> wrote:
        >>
        >>     Daniel,
        >>
        >>     your link seems not reachable.
        >>     We'll wait for the wiki page :).
        >>
        >>     Btw, much appreciated, thanks!
        >>
        >>     -giovanni
        >>
        >>     On Mon, May 17, 2010 at 4:54 PM, Michael S  Collins
        >> <msc at freeswitch.org>
        > wrote:
        >>     > Definitely! Create a wiki page named Kamailio and add your content
        >>     > there. We will then link to it from other appropriate pages.
        >>     >
        >>     > Thanks!
        >>     > -MC (IRC:mercutioviz)
        >>     >
        >>     > Sent from my iPhone
        >>     >
        >>     > On May 17, 2010, at 2:01 AM, Daniel-Constantin Mierla
        >>     >
        >>     > <miconda at gmail.com> wrote:
        >>     >> Hello,
        >>     >>
        >>     >> I put together a tutorial about using kamailio (openser) and
        >>     >> freeswtich
        >>     >> together: the proxy takes care of authentication and registration,
        >>     >> freeswitch of media services, here is a link:
        >>     >>
        >>     >>
        >> http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
        >>     >>
        >>     >> Is it ok to upload it to FS wiki so others can add to it?
        >>     >>
        >>     >> Cheers,
        >>     >> Daniel
        >>     >>
        >>     >> --
        >>     >> Daniel-Constantin Mierla
        >>     >> Kamailio (OpenSER) Advanced Training
        >>     >> Miami, Fl, USA - June 21-23, 2010
        >>     >> http://www.asipto.com/index.php/kamailio-advanced-training/
        >>     >>
        >>     >>
        >>     >> _______________________________________________
        >>     >> FreeSWITCH-users mailing list
        >>     >> FreeSWITCH-users at lists.freeswitch.org
        >>     >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        >>     >>
        >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
        >>     >> users
        >>     >> http://www.freeswitch.org
        >>     >
        >>     > _______________________________________________
        >>     > FreeSWITCH-users mailing list
        >>     > FreeSWITCH-users at lists.freeswitch.org
        >>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        >>     >
        >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
        >>     >ers http://www.freeswitch.org
        >>
        >>     --
        >>     Sincerely,
        >>
        >>     Giovanni Maruzzelli
        >>     Cell : +39-347-2665618
        >>
        >>
        >>     _______________________________________________
        >>     FreeSWITCH-users mailing list
        >>     FreeSWITCH-users at lists.freeswitch.org
        >>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        >>
        >>  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        >>  http://www.freeswitch.org
        >>
        >>
        >>
        >>
        >>
        >> ---------------------------------------------------------------------------
        >> ---
        >>
        >>
        >>   _______________________________________________
        >>   FreeSWITCH-users mailing list
        >>   FreeSWITCH-users at lists.freeswitch.org
        >>   http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        >>
        >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        >>   http://www.freeswitch.org
        >>
        >
        >
        > _______________________________________________
        > FreeSWITCH-users mailing list
        > FreeSWITCH-users at lists.freeswitch.org
        > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        > http://www.freeswitch.org


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





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


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




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






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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From gmaruzz at celliax.org  Mon May 17 12:14:33 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 17 May 2010 21:14:33 +0200
Subject: [Freeswitch-users] kamailio and freeswitch integration
In-Reply-To: <AANLkTikc28zn24mmkprrlrUbwyDKshxY0AEPvQo2uFrj@mail.gmail.com>
References: <4BF105DA.2060209@gmail.com>
	<AANLkTin5sPaagA4RglV3JfJ0rTZJy8_FzNwd4IpVx3LJ@mail.gmail.com> 
	<780492B9CF324B359CA987710CA7738D@MOBILEE1705>
	<201005171355.17228.sos@sokhapkin.dyndns.org> 
	<A3BDB256F37F4E468E8268F529E77081@MOBILEE1705>
	<AANLkTikpv0hVHZUgzCR6HsVuR73rmrUnEnOOSXp3pZq6@mail.gmail.com> 
	<500DFBB8420E46DEAE5A58BAFDA5066C@MOBILEE1705>
	<AANLkTikc28zn24mmkprrlrUbwyDKshxY0AEPvQo2uFrj@mail.gmail.com>
Message-ID: <AANLkTikTl3K5tvi8lJxpcM2W9ycVdcWvm0xsjjLqQMSc@mail.gmail.com>

this link works:
http://ngs.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms

On Mon, May 17, 2010 at 9:03 PM, ram <talk2ram at gmail.com> wrote:
> http://kb.asipto.com/freeswitchkamailio-3.0.x-freeswitch-1.0.6d-ms
>
> is this one correct ?
>
> Ram
>
> On Tue, May 18, 2010 at 12:19 AM, Madovsky <infos at madovsky.org> wrote:
>>
>> I think the problem is the " : "? in the link....
>>
>> ----- Original Message -----
>> From: Phillip Jones
>> To: freeswitch-users at lists.freeswitch.org
>> Sent: Monday, May 17, 2010 2:22 PM
>> Subject: Re: [Freeswitch-users] kamailio and freeswitch integration
>> I think their server is up and down. Worked for me when I posted. Does not
>> now. Base URL ok though.
>>
>> On Mon, May 17, 2010 at 2:10 PM, Madovsky <infos at madovsky.org> wrote:
>>>
>>> weird...
>>> from canada (montreal)
>>> not works
>>>
>>> ----- Original Message -----
>>> From: "Sergey Okhapkin" <sos at sokhapkin.dyndns.org>
>>> To: <freeswitch-users at lists.freeswitch.org>
>>> Sent: Monday, May 17, 2010 1:55 PM
>>> Subject: Re: [Freeswitch-users] kamailio and freeswitch integration
>>>
>>>
>>> > The link works fine to me.
>>> >
>>> > On Monday 17 May 2010, Madovsky wrote:
>>> >> doesn't work
>>> >> ? ----- Original Message -----
>>> >> ? From: Phillip Jones
>>> >> ? To: freeswitch-users at lists.freeswitch.org
>>> >> ? Sent: Monday, May 17, 2010 1:19 PM
>>> >> ? Subject: Re: [Freeswitch-users] kamailio and freeswitch integration
>>> >>
>>> >>
>>> >> ? For those that cannot wait (like me!)
>>> >>
>>> >> ? http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
>>> >>
>>> >>
>>> >> ? On Mon, May 17, 2010 at 12:48 PM, Giovanni Maruzzelli
>>> >> ?<gmaruzz at celliax.org> wrote:
>>> >>
>>> >> ? ? Daniel,
>>> >>
>>> >> ? ? your link seems not reachable.
>>> >> ? ? We'll wait for the wiki page :).
>>> >>
>>> >> ? ? Btw, much appreciated, thanks!
>>> >>
>>> >> ? ? -giovanni
>>> >>
>>> >> ? ? On Mon, May 17, 2010 at 4:54 PM, Michael S ?Collins
>>> >> <msc at freeswitch.org>
>>> > wrote:
>>> >> ? ? > Definitely! Create a wiki page named Kamailio and add your
>>> >> content
>>> >> ? ? > there. We will then link to it from other appropriate pages.
>>> >> ? ? >
>>> >> ? ? > Thanks!
>>> >> ? ? > -MC (IRC:mercutioviz)
>>> >> ? ? >
>>> >> ? ? > Sent from my iPhone
>>> >> ? ? >
>>> >> ? ? > On May 17, 2010, at 2:01 AM, Daniel-Constantin Mierla
>>> >> ? ? >
>>> >> ? ? > <miconda at gmail.com> wrote:
>>> >> ? ? >> Hello,
>>> >> ? ? >>
>>> >> ? ? >> I put together a tutorial about using kamailio (openser) and
>>> >> ? ? >> freeswtich
>>> >> ? ? >> together: the proxy takes care of authentication and
>>> >> registration,
>>> >> ? ? >> freeswitch of media services, here is a link:
>>> >> ? ? >>
>>> >> ? ? >>
>>> >> http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
>>> >> ? ? >>
>>> >> ? ? >> Is it ok to upload it to FS wiki so others can add to it?
>>> >> ? ? >>
>>> >> ? ? >> Cheers,
>>> >> ? ? >> Daniel
>>> >> ? ? >>
>>> >> ? ? >> --
>>> >> ? ? >> Daniel-Constantin Mierla
>>> >> ? ? >> Kamailio (OpenSER) Advanced Training
>>> >> ? ? >> Miami, Fl, USA - June 21-23, 2010
>>> >> ? ? >> http://www.asipto.com/index.php/kamailio-advanced-training/
>>> >> ? ? >>
>>> >> ? ? >>
>>> >> ? ? >> _______________________________________________
>>> >> ? ? >> FreeSWITCH-users mailing list
>>> >> ? ? >> FreeSWITCH-users at lists.freeswitch.org
>>> >> ? ? >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> ? ? >>
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> >> ? ? >> users
>>> >> ? ? >> http://www.freeswitch.org
>>> >> ? ? >
>>> >> ? ? > _______________________________________________
>>> >> ? ? > FreeSWITCH-users mailing list
>>> >> ? ? > FreeSWITCH-users at lists.freeswitch.org
>>> >> ? ? > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> ? ? >
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
>>> >> ? ? >ers http://www.freeswitch.org
>>> >>
>>> >> ? ? --
>>> >> ? ? Sincerely,
>>> >>
>>> >> ? ? Giovanni Maruzzelli
>>> >> ? ? Cell : +39-347-2665618
>>> >>
>>> >>
>>> >> ? ? _______________________________________________
>>> >> ? ? FreeSWITCH-users mailing list
>>> >> ? ? FreeSWITCH-users at lists.freeswitch.org
>>> >> ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>
>>> >>
>>> >> ?UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> ?http://www.freeswitch.org
>>> >>
>>> >>
>>> >>
>>> >>
>>> >>
>>> >>
>>> >> ---------------------------------------------------------------------------
>>> >> ---
>>> >>
>>> >>
>>> >> ? _______________________________________________
>>> >> ? FreeSWITCH-users mailing list
>>> >> ? FreeSWITCH-users at lists.freeswitch.org
>>> >> ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>
>>> >>
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> ? http://www.freeswitch.org
>>> >>
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> ________________________________
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From infos at madovsky.org  Mon May 17 12:15:28 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 17 May 2010 15:15:28 -0400
Subject: [Freeswitch-users] kamailio and freeswitch integration
References: <4BF105DA.2060209@gmail.com><AANLkTin5sPaagA4RglV3JfJ0rTZJy8_FzNwd4IpVx3LJ@mail.gmail.com><780492B9CF324B359CA987710CA7738D@MOBILEE1705><201005171355.17228.sos@sokhapkin.dyndns.org><A3BDB256F37F4E468E8268F529E77081@MOBILEE1705>	<AANLkTikpv0hVHZUgzCR6HsVuR73rmrUnEnOOSXp3pZq6@mail.gmail.com><500DFBB8420E46DEAE5A58BAFDA5066C@MOBILEE1705>
	<4BF19344.4090607@gmail.com>
Message-ID: <D9C27CCF5869493DB6819C75CE67FC46@MOBILEE1705>

works, yeah ! :D
  ----- Original Message ----- 
  From: Daniel-Constantin Mierla 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, May 17, 2010 3:04 PM
  Subject: Re: [Freeswitch-users] kamailio and freeswitch integration


  Hello,

  sorry for inconveniences, but dns propagation really sucks big this time -- record is more than one week old and seems some parts of the world still have an older cache. Being in travel to sipit26 cannot invest much time in it and copy to fs wiki, so i just redirected one of older records to same site, use this link for a while:

  http://ngs.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms

  Colon - ':' -  in link is not a problem.

  @Phillip Jones - probably you got exactly some restart of the web server while redirecting the subdomain, site is up 99.999%.

  Cheers,
  Daniel


  On 5/17/10 8:49 PM, Madovsky wrote: 
    I think the problem is the " : "  in the link....
      ----- Original Message ----- 
      From: Phillip Jones 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Monday, May 17, 2010 2:22 PM
      Subject: Re: [Freeswitch-users] kamailio and freeswitch integration


      I think their server is up and down. Worked for me when I posted. Does not now. Base URL ok though. 


      On Mon, May 17, 2010 at 2:10 PM, Madovsky <infos at madovsky.org> wrote:

        weird...
        from canada (montreal)
        not works


        ----- Original Message -----
        From: "Sergey Okhapkin" <sos at sokhapkin.dyndns.org>
        To: <freeswitch-users at lists.freeswitch.org>

        Sent: Monday, May 17, 2010 1:55 PM
        Subject: Re: [Freeswitch-users] kamailio and freeswitch integration


        > The link works fine to me.
        >
        > On Monday 17 May 2010, Madovsky wrote:
        >> doesn't work
        >>   ----- Original Message -----
        >>   From: Phillip Jones
        >>   To: freeswitch-users at lists.freeswitch.org
        >>   Sent: Monday, May 17, 2010 1:19 PM
        >>   Subject: Re: [Freeswitch-users] kamailio and freeswitch integration
        >>
        >>
        >>   For those that cannot wait (like me!)
        >>
        >>   http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
        >>
        >>
        >>   On Mon, May 17, 2010 at 12:48 PM, Giovanni Maruzzelli
        >>  <gmaruzz at celliax.org> wrote:
        >>
        >>     Daniel,
        >>
        >>     your link seems not reachable.
        >>     We'll wait for the wiki page :).
        >>
        >>     Btw, much appreciated, thanks!
        >>
        >>     -giovanni
        >>
        >>     On Mon, May 17, 2010 at 4:54 PM, Michael S  Collins
        >> <msc at freeswitch.org>
        > wrote:
        >>     > Definitely! Create a wiki page named Kamailio and add your content
        >>     > there. We will then link to it from other appropriate pages.
        >>     >
        >>     > Thanks!
        >>     > -MC (IRC:mercutioviz)
        >>     >
        >>     > Sent from my iPhone
        >>     >
        >>     > On May 17, 2010, at 2:01 AM, Daniel-Constantin Mierla
        >>     >
        >>     > <miconda at gmail.com> wrote:
        >>     >> Hello,
        >>     >>
        >>     >> I put together a tutorial about using kamailio (openser) and
        >>     >> freeswtich
        >>     >> together: the proxy takes care of authentication and registration,
        >>     >> freeswitch of media services, here is a link:
        >>     >>
        >>     >>
        >> http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
        >>     >>
        >>     >> Is it ok to upload it to FS wiki so others can add to it?
        >>     >>
        >>     >> Cheers,
        >>     >> Daniel
        >>     >>
        >>     >> --
        >>     >> Daniel-Constantin Mierla
        >>     >> Kamailio (OpenSER) Advanced Training
        >>     >> Miami, Fl, USA - June 21-23, 2010
        >>     >> http://www.asipto.com/index.php/kamailio-advanced-training/
        >>     >>
        >>     >>
        >>     >> _______________________________________________
        >>     >> FreeSWITCH-users mailing list
        >>     >> FreeSWITCH-users at lists.freeswitch.org
        >>     >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        >>     >>
        >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
        >>     >> users
        >>     >> http://www.freeswitch.org
        >>     >
        >>     > _______________________________________________
        >>     > FreeSWITCH-users mailing list
        >>     > FreeSWITCH-users at lists.freeswitch.org
        >>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        >>     >
        >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
        >>     >ers http://www.freeswitch.org
        >>
        >>     --
        >>     Sincerely,
        >>
        >>     Giovanni Maruzzelli
        >>     Cell : +39-347-2665618
        >>
        >>
        >>     _______________________________________________
        >>     FreeSWITCH-users mailing list
        >>     FreeSWITCH-users at lists.freeswitch.org
        >>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        >>
        >>  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        >>  http://www.freeswitch.org
        >>
        >>
        >>
        >>
        >>
        >> ---------------------------------------------------------------------------
        >> ---
        >>
        >>
        >>   _______________________________________________
        >>   FreeSWITCH-users mailing list
        >>   FreeSWITCH-users at lists.freeswitch.org
        >>   http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        >>
        >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        >>   http://www.freeswitch.org
        >>
        >
        >
        > _______________________________________________
        > FreeSWITCH-users mailing list
        > FreeSWITCH-users at lists.freeswitch.org
        > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
        > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
        > http://www.freeswitch.org


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





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


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

-- 
Daniel-Constantin Mierla
Kamailio (OpenSER) Advanced Training
Miami, Fl, USA - June 21-23, 2010
http://www.asipto.com/index.php/kamailio-advanced-training/


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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From talk2ram at gmail.com  Mon May 17 12:20:13 2010
From: talk2ram at gmail.com (ram)
Date: Tue, 18 May 2010 00:50:13 +0530
Subject: [Freeswitch-users] kamailio and freeswitch integration
In-Reply-To: <4BF19344.4090607@gmail.com>
References: <4BF105DA.2060209@gmail.com>
	<AANLkTin5sPaagA4RglV3JfJ0rTZJy8_FzNwd4IpVx3LJ@mail.gmail.com>
	<780492B9CF324B359CA987710CA7738D@MOBILEE1705>
	<201005171355.17228.sos@sokhapkin.dyndns.org>
	<A3BDB256F37F4E468E8268F529E77081@MOBILEE1705>
	<AANLkTikpv0hVHZUgzCR6HsVuR73rmrUnEnOOSXp3pZq6@mail.gmail.com>
	<500DFBB8420E46DEAE5A58BAFDA5066C@MOBILEE1705>
	<4BF19344.4090607@gmail.com>
Message-ID: <AANLkTinjz-hWcjKP6ANzcIHsnp0Ym1nTAEGA1Pl4bFS_@mail.gmail.com>

it works great



On Tue, May 18, 2010 at 12:34 AM, Daniel-Constantin Mierla <
miconda at gmail.com> wrote:

> Hello,
>
> sorry for inconveniences, but dns propagation really sucks big this time --
> record is more than one week old and seems some parts of the world still
> have an older cache. Being in travel to sipit26 cannot invest much time in
> it and copy to fs wiki, so i just redirected one of older records to same
> site, use this link for a while:
>
> http://ngs.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
>
> Colon - ':' -  in link is not a problem.
>
> @Phillip Jones - probably you got exactly some restart of the web server
> while redirecting the subdomain, site is up 99.999%.
>
> Cheers,
> Daniel
>
>
>
> On 5/17/10 8:49 PM, Madovsky wrote:
>
> I think the problem is the " : "  in the link....
>
> ----- Original Message -----
> *From:* Phillip Jones <pjintheusa at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Monday, May 17, 2010 2:22 PM
> *Subject:* Re: [Freeswitch-users] kamailio and freeswitch integration
>
> I think their server is up and down. Worked for me when I posted. Does not
> now. Base URL ok though.
>
> On Mon, May 17, 2010 at 2:10 PM, Madovsky <infos at madovsky.org> wrote:
>
>> weird...
>> from canada (montreal)
>> not works
>>
>> ----- Original Message -----
>> From: "Sergey Okhapkin" <sos at sokhapkin.dyndns.org>
>> To: <freeswitch-users at lists.freeswitch.org>
>>  Sent: Monday, May 17, 2010 1:55 PM
>> Subject: Re: [Freeswitch-users] kamailio and freeswitch integration
>>
>>
>> > The link works fine to me.
>> >
>> > On Monday 17 May 2010, Madovsky wrote:
>> >> doesn't work
>> >>   ----- Original Message -----
>> >>   From: Phillip Jones
>> >>   To: freeswitch-users at lists.freeswitch.org
>> >>   Sent: Monday, May 17, 2010 1:19 PM
>> >>   Subject: Re: [Freeswitch-users] kamailio and freeswitch integration
>> >>
>> >>
>> >>   For those that cannot wait (like me!)
>> >>
>> >>   http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
>> >>
>> >>
>> >>   On Mon, May 17, 2010 at 12:48 PM, Giovanni Maruzzelli
>> >>  <gmaruzz at celliax.org> wrote:
>> >>
>> >>     Daniel,
>> >>
>> >>     your link seems not reachable.
>> >>     We'll wait for the wiki page :).
>> >>
>> >>     Btw, much appreciated, thanks!
>> >>
>> >>     -giovanni
>> >>
>> >>     On Mon, May 17, 2010 at 4:54 PM, Michael S  Collins
>> >> <msc at freeswitch.org>
>> > wrote:
>> >>     > Definitely! Create a wiki page named Kamailio and add your
>> content
>> >>     > there. We will then link to it from other appropriate pages.
>> >>     >
>> >>     > Thanks!
>> >>     > -MC (IRC:mercutioviz)
>> >>     >
>> >>     > Sent from my iPhone
>> >>     >
>> >>     > On May 17, 2010, at 2:01 AM, Daniel-Constantin Mierla
>> >>     >
>> >>     > <miconda at gmail.com> wrote:
>> >>     >> Hello,
>> >>     >>
>> >>     >> I put together a tutorial about using kamailio (openser) and
>> >>     >> freeswtich
>> >>     >> together: the proxy takes care of authentication and
>> registration,
>> >>     >> freeswitch of media services, here is a link:
>> >>     >>
>> >>     >>
>> >> http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
>> >>     >>
>> >>     >> Is it ok to upload it to FS wiki so others can add to it?
>> >>     >>
>> >>     >> Cheers,
>> >>     >> Daniel
>> >>     >>
>> >>     >> --
>> >>     >> Daniel-Constantin Mierla
>> >>     >> Kamailio (OpenSER) Advanced Training
>> >>     >> Miami, Fl, USA - June 21-23, 2010
>> >>     >> http://www.asipto.com/index.php/kamailio-advanced-training/
>> >>     >>
>> >>     >>
>> >>     >> _______________________________________________
>> >>     >> FreeSWITCH-users mailing list
>> >>     >> FreeSWITCH-users at lists.freeswitch.org
>> >>     >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>     >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> >>     >> users
>> >>     >> http://www.freeswitch.org
>> >>     >
>> >>     > _______________________________________________
>> >>     > FreeSWITCH-users mailing list
>> >>     > FreeSWITCH-users at lists.freeswitch.org
>> >>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>     >
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
>> >>     >ers http://www.freeswitch.org
>> >>
>> >>     --
>> >>     Sincerely,
>> >>
>> >>     Giovanni Maruzzelli
>> >>     Cell : +39-347-2665618
>> >>
>> >>
>> >>     _______________________________________________
>> >>     FreeSWITCH-users mailing list
>> >>     FreeSWITCH-users at lists.freeswitch.org
>> >>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >>  UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>  http://www.freeswitch.org
>> >>
>> >>
>> >>
>> >>
>> >>
>> >>
>> ---------------------------------------------------------------------------
>> >> ---
>> >>
>> >>
>> >>   _______________________________________________
>> >>   FreeSWITCH-users mailing list
>> >>   FreeSWITCH-users at lists.freeswitch.org
>> >>   http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>   http://www.freeswitch.org
>> >>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>  ------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> --
> Daniel-Constantin Mierla
> Kamailio (OpenSER) Advanced Training
> Miami, Fl, USA - June 21-23, 2010
> http://www.asipto.com/index.php/kamailio-advanced-training/
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From miconda at gmail.com  Mon May 17 12:23:54 2010
From: miconda at gmail.com (Daniel-Constantin Mierla)
Date: Mon, 17 May 2010 21:23:54 +0200
Subject: [Freeswitch-users] kamailio and freeswitch integration
In-Reply-To: <5620E34753674B9DACDC4898A9A2EE4F@MOBILEE1705>
References: <4BF105DA.2060209@gmail.com><AANLkTin5sPaagA4RglV3JfJ0rTZJy8_FzNwd4IpVx3LJ@mail.gmail.com><780492B9CF324B359CA987710CA7738D@MOBILEE1705><201005171355.17228.sos@sokhapkin.dyndns.org><A3BDB256F37F4E468E8268F529E77081@MOBILEE1705><AANLkTikpv0hVHZUgzCR6HsVuR73rmrUnEnOOSXp3pZq6@mail.gmail.com><500DFBB8420E46DEAE5A58BAFDA5066C@MOBILEE1705>	<AANLkTikc28zn24mmkprrlrUbwyDKshxY0AEPvQo2uFrj@mail.gmail.com>
	<5620E34753674B9DACDC4898A9A2EE4F@MOBILEE1705>
Message-ID: <4BF197CA.70307@gmail.com>

the link posted by ram is incorrect, it misses the semicolon, anyhow, 
use this one:

http://ngs.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms

Cheers,
Daniel

On 5/17/10 9:14 PM, Madovsky wrote:
> neither :(
>
>     ----- Original Message -----
>     *From:* ram <mailto:talk2ram at gmail.com>
>     *To:* freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>
>     *Sent:* Monday, May 17, 2010 3:03 PM
>     *Subject:* Re: [Freeswitch-users] kamailio and freeswitch integration
>
>     http://kb.asipto.com/freeswitchkamailio-3.0.x-freeswitch-1.0.6d-ms
>     is this one correct ?
>     Ram
>
>     On Tue, May 18, 2010 at 12:19 AM, Madovsky <infos at madovsky.org
>     <mailto:infos at madovsky.org>> wrote:
>
>         I think the problem is the " : "  in the link....
>
>             ----- Original Message -----
>             *From:* Phillip Jones <mailto:pjintheusa at gmail.com>
>             *To:* freeswitch-users at lists.freeswitch.org
>             <mailto:freeswitch-users at lists.freeswitch.org>
>             *Sent:* Monday, May 17, 2010 2:22 PM
>             *Subject:* Re: [Freeswitch-users] kamailio and freeswitch
>             integration
>
>             I think their server is up and down. Worked for me when I
>             posted. Does not now. Base URL ok though.
>
>             On Mon, May 17, 2010 at 2:10 PM, Madovsky
>             <infos at madovsky.org <mailto:infos at madovsky.org>> wrote:
>
>                 weird...
>                 from canada (montreal)
>                 not works
>
>                 ----- Original Message -----
>                 From: "Sergey Okhapkin" <sos at sokhapkin.dyndns.org
>                 <mailto:sos at sokhapkin.dyndns.org>>
>                 To: <freeswitch-users at lists.freeswitch.org
>                 <mailto:freeswitch-users at lists.freeswitch.org>>
>                 Sent: Monday, May 17, 2010 1:55 PM
>                 Subject: Re: [Freeswitch-users] kamailio and
>                 freeswitch integration
>
>
>                 > The link works fine to me.
>                 >
>                 > On Monday 17 May 2010, Madovsky wrote:
>                 >> doesn't work
>                 >>   ----- Original Message -----
>                 >>   From: Phillip Jones
>                 >>   To: freeswitch-users at lists.freeswitch.org
>                 <mailto:freeswitch-users at lists.freeswitch.org>
>                 >>   Sent: Monday, May 17, 2010 1:19 PM
>                 >>   Subject: Re: [Freeswitch-users] kamailio and
>                 freeswitch integration
>                 >>
>                 >>
>                 >>   For those that cannot wait (like me!)
>                 >>
>                 >>
>                 http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
>                 >>
>                 >>
>                 >>   On Mon, May 17, 2010 at 12:48 PM, Giovanni Maruzzelli
>                 >> <gmaruzz at celliax.org <mailto:gmaruzz at celliax.org>>
>                 wrote:
>                 >>
>                 >>     Daniel,
>                 >>
>                 >>     your link seems not reachable.
>                 >>     We'll wait for the wiki page :).
>                 >>
>                 >>     Btw, much appreciated, thanks!
>                 >>
>                 >>     -giovanni
>                 >>
>                 >>     On Mon, May 17, 2010 at 4:54 PM, Michael S  Collins
>                 >> <msc at freeswitch.org <mailto:msc at freeswitch.org>>
>                 > wrote:
>                 >> > Definitely! Create a wiki page named Kamailio and
>                 add your content
>                 >> > there. We will then link to it from other
>                 appropriate pages.
>                 >> >
>                 >> > Thanks!
>                 >> > -MC (IRC:mercutioviz)
>                 >> >
>                 >> > Sent from my iPhone
>                 >> >
>                 >> > On May 17, 2010, at 2:01 AM, Daniel-Constantin Mierla
>                 >> >
>                 >> > <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>                 >> >> Hello,
>                 >> >>
>                 >> >> I put together a tutorial about using kamailio
>                 (openser) and
>                 >> >> freeswtich
>                 >> >> together: the proxy takes care of authentication
>                 and registration,
>                 >> >> freeswitch of media services, here is a link:
>                 >> >>
>                 >> >>
>                 >>
>                 http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
>                 >> >>
>                 >> >> Is it ok to upload it to FS wiki so others can
>                 add to it?
>                 >> >>
>                 >> >> Cheers,
>                 >> >> Daniel
>                 >> >>
>                 >> >> --
>                 >> >> Daniel-Constantin Mierla
>                 >> >> Kamailio (OpenSER) Advanced Training
>                 >> >> Miami, Fl, USA - June 21-23, 2010
>                 >> >>
>                 http://www.asipto.com/index.php/kamailio-advanced-training/
>                 >> >>
>                 >> >>
>                 >> >> _______________________________________________
>                 >> >> FreeSWITCH-users mailing list
>                 >> >> FreeSWITCH-users at lists.freeswitch.org
>                 <mailto:FreeSWITCH-users at lists.freeswitch.org>
>                 >> >>
>                 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>                 >> >>
>                 >>
>                 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>                 >> >> users
>                 >> >> http://www.freeswitch.org
>                 <http://www.freeswitch.org/>
>                 >> >
>                 >> > _______________________________________________
>                 >> > FreeSWITCH-users mailing list
>                 >> > FreeSWITCH-users at lists.freeswitch.org
>                 <mailto:FreeSWITCH-users at lists.freeswitch.org>
>                 >> >
>                 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>                 >> >
>                 >>
>                 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
>                 >> >ers http://www.freeswitch.org
>                 <http://www.freeswitch.org/>
>                 >>
>                 >>     --
>                 >>     Sincerely,
>                 >>
>                 >>     Giovanni Maruzzelli
>                 >>     Cell : +39-347-2665618
>                 >>
>                 >>
>                 >>     _______________________________________________
>                 >>     FreeSWITCH-users mailing list
>                 >> FreeSWITCH-users at lists.freeswitch.org
>                 <mailto:FreeSWITCH-users at lists.freeswitch.org>
>                 >>
>                 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>                 >>
>                 >>
>                  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>                 >> http://www.freeswitch.org <http://www.freeswitch.org/>
>                 >>
>                 >>
>                 >>
>                 >>
>                 >>
>                 >>
>                 ---------------------------------------------------------------------------
>                 >> ---
>                 >>
>                 >>
>                 >>   _______________________________________________
>                 >>   FreeSWITCH-users mailing list
>                 >> FreeSWITCH-users at lists.freeswitch.org
>                 <mailto:FreeSWITCH-users at lists.freeswitch.org>
>                 >>
>                 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>                 >>
>                 >>
>                 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>                 >> http://www.freeswitch.org <http://www.freeswitch.org/>
>                 >>
>                 >
>                 >
>                 > _______________________________________________
>                 > FreeSWITCH-users mailing list
>                 > FreeSWITCH-users at lists.freeswitch.org
>                 <mailto:FreeSWITCH-users at lists.freeswitch.org>
>                 >
>                 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>                 >
>                 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>                 > http://www.freeswitch.org <http://www.freeswitch.org/>
>
>
>                 _______________________________________________
>                 FreeSWITCH-users mailing list
>                 FreeSWITCH-users at lists.freeswitch.org
>                 <mailto:FreeSWITCH-users at lists.freeswitch.org>
>                 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>                 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>                 http://www.freeswitch.org <http://www.freeswitch.org/>
>
>
>             ------------------------------------------------------------------------
>             _______________________________________________
>             FreeSWITCH-users mailing list
>             FreeSWITCH-users at lists.freeswitch.org
>             <mailto:FreeSWITCH-users at lists.freeswitch.org>
>             http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>             UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>             http://www.freeswitch.org <http://www.freeswitch.org/>
>
>
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org <http://www.freeswitch.org/>
>
>
>     ------------------------------------------------------------------------
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    

-- 
Daniel-Constantin Mierla
Kamailio (OpenSER) Advanced Training
Miami, Fl, USA - June 21-23, 2010
http://www.asipto.com/index.php/kamailio-advanced-training/

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From talk2ram at gmail.com  Mon May 17 12:39:45 2010
From: talk2ram at gmail.com (ram)
Date: Tue, 18 May 2010 01:09:45 +0530
Subject: [Freeswitch-users] kamailio and freeswitch integration
In-Reply-To: <4BF197CA.70307@gmail.com>
References: <4BF105DA.2060209@gmail.com>
	<AANLkTin5sPaagA4RglV3JfJ0rTZJy8_FzNwd4IpVx3LJ@mail.gmail.com>
	<780492B9CF324B359CA987710CA7738D@MOBILEE1705>
	<201005171355.17228.sos@sokhapkin.dyndns.org>
	<A3BDB256F37F4E468E8268F529E77081@MOBILEE1705>
	<AANLkTikpv0hVHZUgzCR6HsVuR73rmrUnEnOOSXp3pZq6@mail.gmail.com>
	<500DFBB8420E46DEAE5A58BAFDA5066C@MOBILEE1705>
	<AANLkTikc28zn24mmkprrlrUbwyDKshxY0AEPvQo2uFrj@mail.gmail.com>
	<5620E34753674B9DACDC4898A9A2EE4F@MOBILEE1705>
	<4BF197CA.70307@gmail.com>
Message-ID: <AANLkTimjni4cvbLx2l_XM0pAKEuHJp6DHv2ImPgla4oj@mail.gmail.com>

the below link works



On Tue, May 18, 2010 at 12:53 AM, Daniel-Constantin Mierla <
miconda at gmail.com> wrote:

> the link posted by ram is incorrect, it misses the semicolon, anyhow, use
> this one:
>
> http://ngs.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
>
> Cheers,
> Daniel
>
>
> On 5/17/10 9:14 PM, Madovsky wrote:
>
> neither :(
>
> ----- Original Message -----
> *From:* ram <talk2ram at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Monday, May 17, 2010 3:03 PM
> *Subject:* Re: [Freeswitch-users] kamailio and freeswitch integration
>
> http://kb.asipto.com/freeswitchkamailio-3.0.x-freeswitch-1.0.6d-ms
>
> is this one correct ?
>
> Ram
>
> On Tue, May 18, 2010 at 12:19 AM, Madovsky <infos at madovsky.org> wrote:
>
>>  I think the problem is the " : "  in the link....
>>
>>  ----- Original Message -----
>> *From:* Phillip Jones <pjintheusa at gmail.com>
>> *To:* freeswitch-users at lists.freeswitch.org
>>  *Sent:* Monday, May 17, 2010 2:22 PM
>> *Subject:* Re: [Freeswitch-users] kamailio and freeswitch integration
>>
>> I think their server is up and down. Worked for me when I posted. Does not
>> now. Base URL ok though.
>>
>> On Mon, May 17, 2010 at 2:10 PM, Madovsky <infos at madovsky.org> wrote:
>>
>>> weird...
>>> from canada (montreal)
>>> not works
>>>
>>> ----- Original Message -----
>>> From: "Sergey Okhapkin" <sos at sokhapkin.dyndns.org>
>>> To: <freeswitch-users at lists.freeswitch.org>
>>>  Sent: Monday, May 17, 2010 1:55 PM
>>> Subject: Re: [Freeswitch-users] kamailio and freeswitch integration
>>>
>>>
>>> > The link works fine to me.
>>> >
>>> > On Monday 17 May 2010, Madovsky wrote:
>>> >> doesn't work
>>> >>   ----- Original Message -----
>>> >>   From: Phillip Jones
>>> >>   To: freeswitch-users at lists.freeswitch.org
>>> >>   Sent: Monday, May 17, 2010 1:19 PM
>>> >>   Subject: Re: [Freeswitch-users] kamailio and freeswitch integration
>>> >>
>>> >>
>>> >>   For those that cannot wait (like me!)
>>> >>
>>> >>   http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
>>> >>
>>> >>
>>> >>   On Mon, May 17, 2010 at 12:48 PM, Giovanni Maruzzelli
>>> >>  <gmaruzz at celliax.org> wrote:
>>> >>
>>> >>     Daniel,
>>> >>
>>> >>     your link seems not reachable.
>>> >>     We'll wait for the wiki page :).
>>> >>
>>> >>     Btw, much appreciated, thanks!
>>> >>
>>> >>     -giovanni
>>> >>
>>> >>     On Mon, May 17, 2010 at 4:54 PM, Michael S  Collins
>>> >> <msc at freeswitch.org>
>>> > wrote:
>>> >>     > Definitely! Create a wiki page named Kamailio and add your
>>> content
>>> >>     > there. We will then link to it from other appropriate pages.
>>> >>     >
>>> >>     > Thanks!
>>> >>     > -MC (IRC:mercutioviz)
>>> >>     >
>>> >>     > Sent from my iPhone
>>> >>     >
>>> >>     > On May 17, 2010, at 2:01 AM, Daniel-Constantin Mierla
>>> >>     >
>>> >>     > <miconda at gmail.com> wrote:
>>> >>     >> Hello,
>>> >>     >>
>>> >>     >> I put together a tutorial about using kamailio (openser) and
>>> >>     >> freeswtich
>>> >>     >> together: the proxy takes care of authentication and
>>> registration,
>>> >>     >> freeswitch of media services, here is a link:
>>> >>     >>
>>> >>     >>
>>> >> http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
>>> >>     >>
>>> >>     >> Is it ok to upload it to FS wiki so others can add to it?
>>> >>     >>
>>> >>     >> Cheers,
>>> >>     >> Daniel
>>> >>     >>
>>> >>     >> --
>>> >>     >> Daniel-Constantin Mierla
>>> >>     >> Kamailio (OpenSER) Advanced Training
>>> >>     >> Miami, Fl, USA - June 21-23, 2010
>>> >>     >> http://www.asipto.com/index.php/kamailio-advanced-training/
>>> >>     >>
>>> >>     >>
>>> >>     >> _______________________________________________
>>> >>     >> FreeSWITCH-users mailing list
>>> >>     >> FreeSWITCH-users at lists.freeswitch.org
>>> >>     >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>     >>
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> >>     >> users
>>> >>     >> http://www.freeswitch.org
>>> >>     >
>>> >>     > _______________________________________________
>>> >>     > FreeSWITCH-users mailing list
>>> >>     > FreeSWITCH-users at lists.freeswitch.org
>>> >>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>     >
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
>>> >>     >ers http://www.freeswitch.org
>>> >>
>>> >>     --
>>> >>     Sincerely,
>>> >>
>>> >>     Giovanni Maruzzelli
>>> >>     Cell : +39-347-2665618
>>> >>
>>> >>
>>> >>     _______________________________________________
>>> >>     FreeSWITCH-users mailing list
>>> >>     FreeSWITCH-users at lists.freeswitch.org
>>> >>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>
>>> >>  UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>  http://www.freeswitch.org
>>> >>
>>> >>
>>> >>
>>> >>
>>> >>
>>> >>
>>> ---------------------------------------------------------------------------
>>> >> ---
>>> >>
>>> >>
>>> >>   _______________________________________________
>>> >>   FreeSWITCH-users mailing list
>>> >>   FreeSWITCH-users at lists.freeswitch.org
>>> >>   http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>
>>> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>   http://www.freeswitch.org
>>> >>
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>  ------------------------------
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>  ------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> --
> Daniel-Constantin Mierla
> Kamailio (OpenSER) Advanced Training
> Miami, Fl, USA - June 21-23, 2010
> http://www.asipto.com/index.php/kamailio-advanced-training/
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From pjintheusa at gmail.com  Mon May 17 13:20:45 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Mon, 17 May 2010 16:20:45 -0400
Subject: [Freeswitch-users] kamailio and freeswitch integration
In-Reply-To: <AANLkTimjni4cvbLx2l_XM0pAKEuHJp6DHv2ImPgla4oj@mail.gmail.com>
References: <4BF105DA.2060209@gmail.com>
	<780492B9CF324B359CA987710CA7738D@MOBILEE1705>
	<201005171355.17228.sos@sokhapkin.dyndns.org>
	<A3BDB256F37F4E468E8268F529E77081@MOBILEE1705>
	<AANLkTikpv0hVHZUgzCR6HsVuR73rmrUnEnOOSXp3pZq6@mail.gmail.com>
	<500DFBB8420E46DEAE5A58BAFDA5066C@MOBILEE1705>
	<AANLkTikc28zn24mmkprrlrUbwyDKshxY0AEPvQo2uFrj@mail.gmail.com>
	<5620E34753674B9DACDC4898A9A2EE4F@MOBILEE1705>
	<4BF197CA.70307@gmail.com>
	<AANLkTimjni4cvbLx2l_XM0pAKEuHJp6DHv2ImPgla4oj@mail.gmail.com>
Message-ID: <AANLkTimZMvrD-lRzPefON8nsaduD1t37wyl3iWHyaVQj@mail.gmail.com>

Thanks Daniel for posting. It was worth the wait.

On Mon, May 17, 2010 at 3:39 PM, ram <talk2ram at gmail.com> wrote:

> the below link works
>
>
>
> On Tue, May 18, 2010 at 12:53 AM, Daniel-Constantin Mierla <
> miconda at gmail.com> wrote:
>
>> the link posted by ram is incorrect, it misses the semicolon, anyhow, use
>> this one:
>>
>> http://ngs.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
>>
>> Cheers,
>> Daniel
>>
>>
>> On 5/17/10 9:14 PM, Madovsky wrote:
>>
>> neither :(
>>
>> ----- Original Message -----
>> *From:* ram <talk2ram at gmail.com>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Sent:* Monday, May 17, 2010 3:03 PM
>> *Subject:* Re: [Freeswitch-users] kamailio and freeswitch integration
>>
>> http://kb.asipto.com/freeswitchkamailio-3.0.x-freeswitch-1.0.6d-ms
>>
>> is this one correct ?
>>
>> Ram
>>
>> On Tue, May 18, 2010 at 12:19 AM, Madovsky <infos at madovsky.org> wrote:
>>
>>>  I think the problem is the " : "  in the link....
>>>
>>>  ----- Original Message -----
>>> *From:* Phillip Jones <pjintheusa at gmail.com>
>>> *To:* freeswitch-users at lists.freeswitch.org
>>>   *Sent:* Monday, May 17, 2010 2:22 PM
>>> *Subject:* Re: [Freeswitch-users] kamailio and freeswitch integration
>>>
>>> I think their server is up and down. Worked for me when I posted. Does
>>> not now. Base URL ok though.
>>>
>>> On Mon, May 17, 2010 at 2:10 PM, Madovsky <infos at madovsky.org> wrote:
>>>
>>>> weird...
>>>> from canada (montreal)
>>>> not works
>>>>
>>>> ----- Original Message -----
>>>> From: "Sergey Okhapkin" <sos at sokhapkin.dyndns.org>
>>>> To: <freeswitch-users at lists.freeswitch.org>
>>>>  Sent: Monday, May 17, 2010 1:55 PM
>>>> Subject: Re: [Freeswitch-users] kamailio and freeswitch integration
>>>>
>>>>
>>>> > The link works fine to me.
>>>> >
>>>> > On Monday 17 May 2010, Madovsky wrote:
>>>> >> doesn't work
>>>> >>   ----- Original Message -----
>>>> >>   From: Phillip Jones
>>>> >>   To: freeswitch-users at lists.freeswitch.org
>>>> >>   Sent: Monday, May 17, 2010 1:19 PM
>>>> >>   Subject: Re: [Freeswitch-users] kamailio and freeswitch integration
>>>> >>
>>>> >>
>>>> >>   For those that cannot wait (like me!)
>>>> >>
>>>> >>
>>>> http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
>>>> >>
>>>> >>
>>>> >>   On Mon, May 17, 2010 at 12:48 PM, Giovanni Maruzzelli
>>>> >>  <gmaruzz at celliax.org> wrote:
>>>> >>
>>>> >>     Daniel,
>>>> >>
>>>> >>     your link seems not reachable.
>>>> >>     We'll wait for the wiki page :).
>>>> >>
>>>> >>     Btw, much appreciated, thanks!
>>>> >>
>>>> >>     -giovanni
>>>> >>
>>>> >>     On Mon, May 17, 2010 at 4:54 PM, Michael S  Collins
>>>> >> <msc at freeswitch.org>
>>>> > wrote:
>>>> >>     > Definitely! Create a wiki page named Kamailio and add your
>>>> content
>>>> >>     > there. We will then link to it from other appropriate pages.
>>>> >>     >
>>>> >>     > Thanks!
>>>> >>     > -MC (IRC:mercutioviz)
>>>> >>     >
>>>> >>     > Sent from my iPhone
>>>> >>     >
>>>> >>     > On May 17, 2010, at 2:01 AM, Daniel-Constantin Mierla
>>>> >>     >
>>>> >>     > <miconda at gmail.com> wrote:
>>>> >>     >> Hello,
>>>> >>     >>
>>>> >>     >> I put together a tutorial about using kamailio (openser) and
>>>> >>     >> freeswtich
>>>> >>     >> together: the proxy takes care of authentication and
>>>> registration,
>>>> >>     >> freeswitch of media services, here is a link:
>>>> >>     >>
>>>> >>     >>
>>>> >> http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
>>>> >>     >>
>>>> >>     >> Is it ok to upload it to FS wiki so others can add to it?
>>>> >>     >>
>>>> >>     >> Cheers,
>>>> >>     >> Daniel
>>>> >>     >>
>>>> >>     >> --
>>>> >>     >> Daniel-Constantin Mierla
>>>> >>     >> Kamailio (OpenSER) Advanced Training
>>>> >>     >> Miami, Fl, USA - June 21-23, 2010
>>>> >>     >> http://www.asipto.com/index.php/kamailio-advanced-training/
>>>> >>     >>
>>>> >>     >>
>>>> >>     >> _______________________________________________
>>>> >>     >> FreeSWITCH-users mailing list
>>>> >>     >> FreeSWITCH-users at lists.freeswitch.org
>>>> >>     >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >>     >>
>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>>> >>     >> users
>>>> >>     >> http://www.freeswitch.org
>>>> >>     >
>>>> >>     > _______________________________________________
>>>> >>     > FreeSWITCH-users mailing list
>>>> >>     > FreeSWITCH-users at lists.freeswitch.org
>>>> >>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >>     >
>>>> >> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-us
>>>> >>     >ers http://www.freeswitch.org
>>>> >>
>>>> >>     --
>>>> >>     Sincerely,
>>>> >>
>>>> >>     Giovanni Maruzzelli
>>>> >>     Cell : +39-347-2665618
>>>> >>
>>>> >>
>>>> >>     _______________________________________________
>>>> >>     FreeSWITCH-users mailing list
>>>> >>     FreeSWITCH-users at lists.freeswitch.org
>>>> >>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >>
>>>> >>  UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >>  http://www.freeswitch.org
>>>> >>
>>>> >>
>>>> >>
>>>> >>
>>>> >>
>>>> >>
>>>> ---------------------------------------------------------------------------
>>>> >> ---
>>>> >>
>>>> >>
>>>> >>   _______________________________________________
>>>> >>   FreeSWITCH-users mailing list
>>>> >>   FreeSWITCH-users at lists.freeswitch.org
>>>> >>   http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >>
>>>> >> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >>   http://www.freeswitch.org
>>>> >>
>>>> >
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>  ------------------------------
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>  ------------------------------
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>> --
>> Daniel-Constantin Mierla
>> Kamailio (OpenSER) Advanced Training
>> Miami, Fl, USA - June 21-23, 2010
>> http://www.asipto.com/index.php/kamailio-advanced-training/
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon May 17 14:16:11 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 17 May 2010 14:16:11 -0700
Subject: [Freeswitch-users] dll sym error
In-Reply-To: <AANLkTikfEtaFuN9xiVFDEa11yEnpsM7PgdTQ7h5xFqN-@mail.gmail.com>
References: <AANLkTikfEtaFuN9xiVFDEa11yEnpsM7PgdTQ7h5xFqN-@mail.gmail.com>
Message-ID: <AANLkTineH1ZIn4Bd-Ll5IXV0q9cVxM7_RuFnGWy6urm-@mail.gmail.com>

Could you supply more details, please?

On Sat, May 15, 2010 at 3:14 AM, babak yakhchali <babak.freeswitch at gmail.com
> wrote:

> Hi
> I'm changing one of say modules, but when I try to load, it gives "dll sym
> error"?
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon May 17 14:24:58 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 17 May 2010 14:24:58 -0700
Subject: [Freeswitch-users] mod_ivr with condition
In-Reply-To: <AANLkTilT9qs9b61U6w4thmO3KPfMh3PM0VqZRm_l4ryi@mail.gmail.com>
References: <AANLkTilT9qs9b61U6w4thmO3KPfMh3PM0VqZRm_l4ryi@mail.gmail.com>
Message-ID: <AANLkTikV-sqaTQkKvpPmkGvQqXv_xIGTXzZnDAAi0-oY@mail.gmail.com>

When you say you setup IVR successfully does that mean you created a new
<menu> like in conf/ivr_menus.xml? If so, then you can accept digits from
the caller while he is in the IVR. To match exactly "1000" from the caller
do this:
<entry action="menu-exec-app" digits="/^(1000)$/" param="transfer $1 XML
features"/>

You could then have a less specific regex to handle all other input, even if
it's more or less than four digits:
<entry action="menu-exec-app" digits="/^(\d+)$/" param="transfer $1 XML
features"/>

You just need to decide what to do when the caller dials the digits:
menu-exec-app or menu-sub or whatever.

-MC

On Sun, May 16, 2010 at 6:46 AM, budi wibowo <bwibowo at gmail.com> wrote:

> dear all
> i setup ivr successfully following xml sample, i want to add functionality
> to give condition based on calling number,
> generally i want this:
> if calling_id =1000
>  then do someaction
>  else do some other action
>
>
> can i use xml for this or should use javascript ?
>
> regards
>
> budi
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon May 17 14:30:15 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 17 May 2010 14:30:15 -0700
Subject: [Freeswitch-users] Setup help for outbound dialing
In-Reply-To: <AANLkTinjlIr97zIhXgS0LGmjfNOeXWvNObS4gweZZnd_@mail.gmail.com>
References: <AANLkTikGzzmdhr8iHKy39zGdvcNPpsKkBH3snt_6cfDW@mail.gmail.com>
	<AANLkTinjlIr97zIhXgS0LGmjfNOeXWvNObS4gweZZnd_@mail.gmail.com>
Message-ID: <AANLkTilIsSOKpF1FWqAiBfv_akXnmpZfeS9awfhJ_Rg9@mail.gmail.com>

On Sun, May 16, 2010 at 9:55 PM, Bob Coleman <bobc at devassert.com> wrote:

> Hi,
>
> My setup is basically as follows:
>
> Have a sip profile that allows incoming calls(lets call it internal).
> Have setup dialplans to accept the inbound ddi's that I want to allow
> routing to an ivr application.
>
> Our server is hosted inside a sip providers network, so we dont use
> gateways etc, direct sip trunk via internal sofia profile.
>
> I am struggling to setup outbound calling. I can make a call through a
> gateway, but am not sure what to setup to allow an outbound call
> through our internal profile.
>
> eg originate sofia/internal/<phonenumber>@<ipaddress> <extension>
>

That syntax looks basically correct. What happens when you try that? Most
likely you need to turn on SIP debugging and watch the SIP traffic for
clues:
sofia profile internal siptrace on

In any case if you could provide more information we can probably help you
solve the issue.
-MC
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From bobc at devassert.com  Mon May 17 14:49:05 2010
From: bobc at devassert.com (Bob Coleman)
Date: Tue, 18 May 2010 09:49:05 +1200
Subject: [Freeswitch-users] Setup help for outbound dialing
In-Reply-To: <AANLkTilIsSOKpF1FWqAiBfv_akXnmpZfeS9awfhJ_Rg9@mail.gmail.com>
References: <AANLkTikGzzmdhr8iHKy39zGdvcNPpsKkBH3snt_6cfDW@mail.gmail.com>
	<AANLkTinjlIr97zIhXgS0LGmjfNOeXWvNObS4gweZZnd_@mail.gmail.com>
	<AANLkTilIsSOKpF1FWqAiBfv_akXnmpZfeS9awfhJ_Rg9@mail.gmail.com>
Message-ID: <AANLkTinMZKOaC5YfDexsxq-OOjO-0tHt2KGumihGPtN0@mail.gmail.com>

When I try that it tells me it is OK and gives me a uuid for the call,
but the call never leaves the network.

Will turn on the trace so I can provide more info, the fact is if I am
making the call correctly then it probably lies with the sip supplier.

The sip supplier said they expect the originate to be in E.164 format
if that makes any difference.

Thanks

Bob

On Tue, May 18, 2010 at 9:30 AM, Michael Collins <msc at freeswitch.org> wrote:
>
>
> On Sun, May 16, 2010 at 9:55 PM, Bob Coleman <bobc at devassert.com> wrote:
>>
>> Hi,
>>
>> My setup is basically as follows:
>>
>> Have a sip profile that allows incoming calls(lets call it internal).
>> Have setup dialplans to accept the inbound ddi's that I want to allow
>> routing to an ivr application.
>>
>> Our server is hosted inside a sip providers network, so we dont use
>> gateways etc, direct sip trunk via internal sofia profile.
>>
>> I am struggling to setup outbound calling. I can make a call through a
>> gateway, but am not sure what to setup to allow an outbound call
>> through our internal profile.
>>
>> eg originate sofia/internal/<phonenumber>@<ipaddress> <extension>
>
> That syntax looks basically correct. What happens when you try that? Most
> likely you need to turn on SIP debugging and watch the SIP traffic for
> clues:
> sofia profile internal siptrace on
>
> In any case if you could provide more information we can probably help you
> solve the issue.
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From msc at freeswitch.org  Mon May 17 15:06:00 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 17 May 2010 15:06:00 -0700
Subject: [Freeswitch-users] Setup help for outbound dialing
In-Reply-To: <AANLkTinMZKOaC5YfDexsxq-OOjO-0tHt2KGumihGPtN0@mail.gmail.com>
References: <AANLkTikGzzmdhr8iHKy39zGdvcNPpsKkBH3snt_6cfDW@mail.gmail.com>
	<AANLkTinjlIr97zIhXgS0LGmjfNOeXWvNObS4gweZZnd_@mail.gmail.com>
	<AANLkTilIsSOKpF1FWqAiBfv_akXnmpZfeS9awfhJ_Rg9@mail.gmail.com>
	<AANLkTinMZKOaC5YfDexsxq-OOjO-0tHt2KGumihGPtN0@mail.gmail.com>
Message-ID: <AANLkTimD0ORggPc5amdsCM6kTamHE0awp7liERXwTnbf@mail.gmail.com>

On Mon, May 17, 2010 at 2:49 PM, Bob Coleman <bobc at devassert.com> wrote:

> When I try that it tells me it is OK and gives me a uuid for the call,
> but the call never leaves the network.
>
> Will turn on the trace so I can provide more info, the fact is if I am
> making the call correctly then it probably lies with the sip supplier.
>
> The sip supplier said they expect the originate to be in E.164 format
> if that makes any difference.
>
It does. It probably means they want to see +1NxxNxxxxxx so be sure to
include the "+1"

-MC
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From msc at freeswitch.org  Mon May 17 15:08:21 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 17 May 2010 15:08:21 -0700
Subject: [Freeswitch-users] headset for CELT fullband conferencing
In-Reply-To: <4C2CD024.10403@freenet.de>
References: <4C2CD024.10403@freenet.de>
Message-ID: <AANLkTil_KBAK0WUmAg532bPDlV7Spq-n-sU6n96KdhKe@mail.gmail.com>

I suppose it depends on what you want to get out of it, but generally
speaking when a headset is labeled "Skype Certified" it means that it
handles more than the traditional 300Hz to 3300Hz range of typical phones.
Personally I've used two different Logitech USB headsets and I would have to
say that they are both very good. $35 at Wal-Mart is not a bad deal for a
headset that sounds fantastic on CELT. I'm afraid I can't speak to other
headsets.

-MC
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From gmaruzz at celliax.org  Mon May 17 15:29:33 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Tue, 18 May 2010 00:29:33 +0200
Subject: [Freeswitch-users] kamailio and freeswitch integration
In-Reply-To: <AANLkTimZMvrD-lRzPefON8nsaduD1t37wyl3iWHyaVQj@mail.gmail.com>
References: <4BF105DA.2060209@gmail.com>
	<201005171355.17228.sos@sokhapkin.dyndns.org> 
	<A3BDB256F37F4E468E8268F529E77081@MOBILEE1705>
	<AANLkTikpv0hVHZUgzCR6HsVuR73rmrUnEnOOSXp3pZq6@mail.gmail.com> 
	<500DFBB8420E46DEAE5A58BAFDA5066C@MOBILEE1705>
	<AANLkTikc28zn24mmkprrlrUbwyDKshxY0AEPvQo2uFrj@mail.gmail.com> 
	<5620E34753674B9DACDC4898A9A2EE4F@MOBILEE1705>
	<4BF197CA.70307@gmail.com> 
	<AANLkTimjni4cvbLx2l_XM0pAKEuHJp6DHv2ImPgla4oj@mail.gmail.com> 
	<AANLkTimZMvrD-lRzPefON8nsaduD1t37wyl3iWHyaVQj@mail.gmail.com>
Message-ID: <AANLkTin6Tc4guy_8gvLYHq8uB8iiVaRXUFjRRD6hHqbx@mail.gmail.com>

On Mon, May 17, 2010 at 10:20 PM, Phillip Jones <pjintheusa at gmail.com> wrote:
> Thanks Daniel for posting. It was worth the wait.

yep it was, thanks Daniel!
(can we look forward to the next article "dynamic real time database
integration with FS and Kamailio"? ;) )




>
> On Mon, May 17, 2010 at 3:39 PM, ram <talk2ram at gmail.com> wrote:
>>
>> the below link works
>>
>>
>> On Tue, May 18, 2010 at 12:53 AM, Daniel-Constantin Mierla
>> <miconda at gmail.com> wrote:
>>>
>>> the link posted by ram is incorrect, it misses the semicolon, anyhow, use
>>> this one:
>>>
>>> http://ngs.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
>>>
>>> Cheers,
>>> Daniel
>>>
>>> On 5/17/10 9:14 PM, Madovsky wrote:
>>>
>>> neither :(
>>>
>>> ----- Original Message -----
>>> From: ram
>>> To: freeswitch-users at lists.freeswitch.org
>>> Sent: Monday, May 17, 2010 3:03 PM
>>> Subject: Re: [Freeswitch-users] kamailio and freeswitch integration
>>> http://kb.asipto.com/freeswitchkamailio-3.0.x-freeswitch-1.0.6d-ms
>>>
>>> is this one correct ?
>>>
>>> Ram
>>>
>>> On Tue, May 18, 2010 at 12:19 AM, Madovsky <infos at madovsky.org> wrote:
>>>>
>>>> I think the problem is the " : "? in the link....
>>>>
>>>> ----- Original Message -----
>>>> From: Phillip Jones
>>>> To: freeswitch-users at lists.freeswitch.org
>>>> Sent: Monday, May 17, 2010 2:22 PM
>>>> Subject: Re: [Freeswitch-users] kamailio and freeswitch integration
>>>> I think their server is up and down. Worked for me when I posted. Does
>>>> not now. Base URL ok though.
>>>>
>>>> On Mon, May 17, 2010 at 2:10 PM, Madovsky <infos at madovsky.org> wrote:
>>>>>
>>>>> weird...
>>>>> from canada (montreal)
>>>>> not works
>>>>>
>>>>> ----- Original Message -----
>>>>> From: "Sergey Okhapkin" <sos at sokhapkin.dyndns.org>
>>>>> To: <freeswitch-users at lists.freeswitch.org>
>>>>> Sent: Monday, May 17, 2010 1:55 PM
>>>>> Subject: Re: [Freeswitch-users] kamailio and freeswitch integration
>>>>>
>>>>>
>>>>> > The link works fine to me.
>>>>> >
>>>>> > On Monday 17 May 2010, Madovsky wrote:
>>>>> >> doesn't work
>>>>> >> ? ----- Original Message -----
>>>>> >> ? From: Phillip Jones
>>>>> >> ? To: freeswitch-users at lists.freeswitch.org
>>>>> >> ? Sent: Monday, May 17, 2010 1:19 PM
>>>>> >> ? Subject: Re: [Freeswitch-users] kamailio and freeswitch
>>>>> >> integration
>>>>> >>
>>>>> >>
>>>>> >> ? For those that cannot wait (like me!)
>>>>> >>
>>>>> >>
>>>>> >> http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
>>>>> >>
>>>>> >>
>>>>> >> ? On Mon, May 17, 2010 at 12:48 PM, Giovanni Maruzzelli
>>>>> >> ?<gmaruzz at celliax.org> wrote:
>>>>> >>
>>>>> >> ? ? Daniel,
>>>>> >>
>>>>> >> ? ? your link seems not reachable.
>>>>> >> ? ? We'll wait for the wiki page :).
>>>>> >>
>>>>> >> ? ? Btw, much appreciated, thanks!
>>>>> >>
>>>>> >> ? ? -giovanni
>>>>> >>
>>>>> >> ? ? On Mon, May 17, 2010 at 4:54 PM, Michael S ?Collins
>>>>> >> <msc at freeswitch.org>
>>>>> > wrote:
>>>>> >> ? ? > Definitely! Create a wiki page named Kamailio and add your
>>>>> >> content
>>>>> >> ? ? > there. We will then link to it from other appropriate pages.
>>>>> >> ? ? >
>>>>> >> ? ? > Thanks!
>>>>> >> ? ? > -MC (IRC:mercutioviz)
>>>>> >> ? ? >
>>>>> >> ? ? > Sent from my iPhone
>>>>> >> ? ? >
>>>>> >> ? ? > On May 17, 2010, at 2:01 AM, Daniel-Constantin Mierla
>>>>> >> ? ? >
>>>>> >> ? ? > <miconda at gmail.com> wrote:
>>>>> >> ? ? >> Hello,
>>>>> >> ? ? >>
>>>>> >> ? ? >> I put together a tutorial about using kamailio (openser) and
>>>>> >> ? ? >> freeswtich
>>>>> >> ? ? >> together: the proxy takes care of authentication and
>>>>> >> registration,
>>>>> >> ? ? >> freeswitch of media services, here is a link:
>>>>> >> ? ? >>
>>>>> >> ? ? >>
>>>>> >> http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
>>>>> >> ? ? >>
>>>>> >> ? ? >> Is it ok to upload it to FS wiki so others can add to it?
>>>>> >> ? ? >>
>>>>> >> ? ? >> Cheers,
>>>>> >> ? ? >> Daniel
>>>>> >> ? ? >>
>>>>> >> ? ? >> --
>>>>> >> ? ? >> Daniel-Constantin Mierla
>>>>> >> ? ? >> Kamailio (OpenSER) Advanced Training
>>>>> >> ? ? >> Miami, Fl, USA - June 21-23, 2010
>>>>> >> ? ? >> http://www.asipto.com/index.php/kamailio-advanced-training/
>>>>> >> ? ? >>
>>>>> >> ? ? >>
>>>>> >> ? ? >> _______________________________________________
>>>>> >> ? ? >> FreeSWITCH-users mailing list
>>>>> >> ? ? >> FreeSWITCH-users at lists.freeswitch.org
>>>>> >> ? ? >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >> ? ? >>
>>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>> >> ? ? >> users
>>>>> >> ? ? >> http://www.freeswitch.org
>>>>> >> ? ? >
>>>>> >> ? ? > _______________________________________________
>>>>> >> ? ? > FreeSWITCH-users mailing list
>>>>> >> ? ? > FreeSWITCH-users at lists.freeswitch.org
>>>>> >> ? ? > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >> ? ? >
>>>>> >>
>>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
>>>>> >> ? ? >ers http://www.freeswitch.org
>>>>> >>
>>>>> >> ? ? --
>>>>> >> ? ? Sincerely,
>>>>> >>
>>>>> >> ? ? Giovanni Maruzzelli
>>>>> >> ? ? Cell : +39-347-2665618
>>>>> >>
>>>>> >>
>>>>> >> ? ? _______________________________________________
>>>>> >> ? ? FreeSWITCH-users mailing list
>>>>> >> ? ? FreeSWITCH-users at lists.freeswitch.org
>>>>> >> ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >>
>>>>> >>
>>>>> >> ?UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> >> ?http://www.freeswitch.org
>>>>> >>
>>>>> >>
>>>>> >>
>>>>> >>
>>>>> >>
>>>>> >>
>>>>> >> ---------------------------------------------------------------------------
>>>>> >> ---
>>>>> >>
>>>>> >>
>>>>> >> ? _______________________________________________
>>>>> >> ? FreeSWITCH-users mailing list
>>>>> >> ? FreeSWITCH-users at lists.freeswitch.org
>>>>> >> ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >>
>>>>> >>
>>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> >> ? http://www.freeswitch.org
>>>>> >>
>>>>> >
>>>>> >
>>>>> > _______________________________________________
>>>>> > FreeSWITCH-users mailing list
>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >
>>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> > http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>> ________________________________
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> ________________________________
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> --
>>> Daniel-Constantin Mierla
>>> Kamailio (OpenSER) Advanced Training
>>> Miami, Fl, USA - June 21-23, 2010
>>> http://www.asipto.com/index.php/kamailio-advanced-training/
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From anthony.minessale at gmail.com  Mon May 17 17:00:25 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 17 May 2010 19:00:25 -0500
Subject: [Freeswitch-users] EventConsumer and multi-threading
In-Reply-To: <AANLkTinz4WHJrO_6q3anFdtiU3wZyaH8XHaaGFjVpQhB@mail.gmail.com>
References: <AANLkTinz4WHJrO_6q3anFdtiU3wZyaH8XHaaGFjVpQhB@mail.gmail.com>
Message-ID: <AANLkTilKAf1tWJuIrn4WQgQT0lcyX3vIoMNlpejB7fPQ@mail.gmail.com>

if you update to latest the eventConsumer has a flush() method to remove all
events
also it should do this when you destroy the object.


On Sun, May 16, 2010 at 4:07 AM, babak yakhchali <babak.freeswitch at gmail.com
> wrote:

> Hi
> I'm using mod_managed and I've used EventConsumer on a thread to listen for
> incomming events but whenever I wanna reload the .dll file of my script it
> complains about thread aborting. I used a dummy custom:dead event to end the
> thread but next time my dll loaded the custom:dead event is still in events
> and the tread ends. Is there any workaround to flush EventConsumer??
> Thanks
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
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
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From mitch.capper at gmail.com  Mon May 17 18:51:57 2010
From: mitch.capper at gmail.com (Mitch Capper)
Date: Mon, 17 May 2010 21:51:57 -0400
Subject: [Freeswitch-users] headset for CELT fullband conferencing
In-Reply-To: <AANLkTil_KBAK0WUmAg532bPDlV7Spq-n-sU6n96KdhKe@mail.gmail.com>
References: <4C2CD024.10403@freenet.de>
	<AANLkTil_KBAK0WUmAg532bPDlV7Spq-n-sU6n96KdhKe@mail.gmail.com>
Message-ID: <AANLkTilanSqmkLsjveaHlxIyaiTz3wJk-fJ1vi46l0Yo@mail.gmail.com>

The Jabra 9400 series (eg 9470) supports both wide and arrow band
transmission and work well.

~Mitch

On Mon, May 17, 2010 at 6:08 PM, Michael Collins <msc at freeswitch.org> wrote:

> I suppose it depends on what you want to get out of it, but generally
> speaking when a headset is labeled "Skype Certified" it means that it
> handles more than the traditional 300Hz to 3300Hz range of typical phones.
> Personally I've used two different Logitech USB headsets and I would have to
> say that they are both very good. $35 at Wal-Mart is not a bad deal for a
> headset that sounds fantastic on CELT. I'm afraid I can't speak to other
> headsets.
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From talk2ram at gmail.com  Mon May 17 20:23:44 2010
From: talk2ram at gmail.com (ram)
Date: Tue, 18 May 2010 08:53:44 +0530
Subject: [Freeswitch-users] kamailio and freeswitch integration
In-Reply-To: <AANLkTimZMvrD-lRzPefON8nsaduD1t37wyl3iWHyaVQj@mail.gmail.com>
References: <4BF105DA.2060209@gmail.com>
	<201005171355.17228.sos@sokhapkin.dyndns.org>
	<A3BDB256F37F4E468E8268F529E77081@MOBILEE1705>
	<AANLkTikpv0hVHZUgzCR6HsVuR73rmrUnEnOOSXp3pZq6@mail.gmail.com>
	<500DFBB8420E46DEAE5A58BAFDA5066C@MOBILEE1705>
	<AANLkTikc28zn24mmkprrlrUbwyDKshxY0AEPvQo2uFrj@mail.gmail.com>
	<5620E34753674B9DACDC4898A9A2EE4F@MOBILEE1705>
	<4BF197CA.70307@gmail.com>
	<AANLkTimjni4cvbLx2l_XM0pAKEuHJp6DHv2ImPgla4oj@mail.gmail.com>
	<AANLkTimZMvrD-lRzPefON8nsaduD1t37wyl3iWHyaVQj@mail.gmail.com>
Message-ID: <AANLkTimyQcbi7Se_OBtjGd7WCcGq3k0qQ6kQD6PN_6pY@mail.gmail.com>

Hi

I have done realtime with Freeswitch with Mysql
but document not made.. will post the document soon

ram

On Tue, May 18, 2010 at 1:50 AM, Phillip Jones <pjintheusa at gmail.com> wrote:

> Thanks Daniel for posting. It was worth the wait.
>
>
> On Mon, May 17, 2010 at 3:39 PM, ram <talk2ram at gmail.com> wrote:
>
>> the below link works
>>
>>
>>
>> On Tue, May 18, 2010 at 12:53 AM, Daniel-Constantin Mierla <
>> miconda at gmail.com> wrote:
>>
>>> the link posted by ram is incorrect, it misses the semicolon, anyhow, use
>>> this one:
>>>
>>> http://ngs.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
>>>
>>> Cheers,
>>> Daniel
>>>
>>>
>>> On 5/17/10 9:14 PM, Madovsky wrote:
>>>
>>> neither :(
>>>
>>> ----- Original Message -----
>>> *From:* ram <talk2ram at gmail.com>
>>> *To:* freeswitch-users at lists.freeswitch.org
>>> *Sent:* Monday, May 17, 2010 3:03 PM
>>> *Subject:* Re: [Freeswitch-users] kamailio and freeswitch integration
>>>
>>> http://kb.asipto.com/freeswitchkamailio-3.0.x-freeswitch-1.0.6d-ms
>>>
>>> is this one correct ?
>>>
>>> Ram
>>>
>>> On Tue, May 18, 2010 at 12:19 AM, Madovsky <infos at madovsky.org> wrote:
>>>
>>>>  I think the problem is the " : "  in the link....
>>>>
>>>>  ----- Original Message -----
>>>> *From:* Phillip Jones <pjintheusa at gmail.com>
>>>> *To:* freeswitch-users at lists.freeswitch.org
>>>>  *Sent:* Monday, May 17, 2010 2:22 PM
>>>> *Subject:* Re: [Freeswitch-users] kamailio and freeswitch integration
>>>>
>>>> I think their server is up and down. Worked for me when I posted. Does
>>>> not now. Base URL ok though.
>>>>
>>>> On Mon, May 17, 2010 at 2:10 PM, Madovsky <infos at madovsky.org> wrote:
>>>>
>>>>> weird...
>>>>> from canada (montreal)
>>>>> not works
>>>>>
>>>>> ----- Original Message -----
>>>>> From: "Sergey Okhapkin" <sos at sokhapkin.dyndns.org>
>>>>> To: <freeswitch-users at lists.freeswitch.org>
>>>>>  Sent: Monday, May 17, 2010 1:55 PM
>>>>> Subject: Re: [Freeswitch-users] kamailio and freeswitch integration
>>>>>
>>>>>
>>>>> > The link works fine to me.
>>>>> >
>>>>> > On Monday 17 May 2010, Madovsky wrote:
>>>>> >> doesn't work
>>>>> >>   ----- Original Message -----
>>>>> >>   From: Phillip Jones
>>>>> >>   To: freeswitch-users at lists.freeswitch.org
>>>>> >>   Sent: Monday, May 17, 2010 1:19 PM
>>>>> >>   Subject: Re: [Freeswitch-users] kamailio and freeswitch
>>>>> integration
>>>>> >>
>>>>> >>
>>>>> >>   For those that cannot wait (like me!)
>>>>> >>
>>>>> >>
>>>>> http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
>>>>> >>
>>>>> >>
>>>>> >>   On Mon, May 17, 2010 at 12:48 PM, Giovanni Maruzzelli
>>>>> >>  <gmaruzz at celliax.org> wrote:
>>>>> >>
>>>>> >>     Daniel,
>>>>> >>
>>>>> >>     your link seems not reachable.
>>>>> >>     We'll wait for the wiki page :).
>>>>> >>
>>>>> >>     Btw, much appreciated, thanks!
>>>>> >>
>>>>> >>     -giovanni
>>>>> >>
>>>>> >>     On Mon, May 17, 2010 at 4:54 PM, Michael S  Collins
>>>>> >> <msc at freeswitch.org>
>>>>> > wrote:
>>>>> >>     > Definitely! Create a wiki page named Kamailio and add your
>>>>> content
>>>>> >>     > there. We will then link to it from other appropriate pages.
>>>>> >>     >
>>>>> >>     > Thanks!
>>>>> >>     > -MC (IRC:mercutioviz)
>>>>> >>     >
>>>>> >>     > Sent from my iPhone
>>>>> >>     >
>>>>> >>     > On May 17, 2010, at 2:01 AM, Daniel-Constantin Mierla
>>>>> >>     >
>>>>> >>     > <miconda at gmail.com> wrote:
>>>>> >>     >> Hello,
>>>>> >>     >>
>>>>> >>     >> I put together a tutorial about using kamailio (openser) and
>>>>> >>     >> freeswtich
>>>>> >>     >> together: the proxy takes care of authentication and
>>>>> registration,
>>>>> >>     >> freeswitch of media services, here is a link:
>>>>> >>     >>
>>>>> >>     >>
>>>>> >> http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
>>>>> >>     >>
>>>>> >>     >> Is it ok to upload it to FS wiki so others can add to it?
>>>>> >>     >>
>>>>> >>     >> Cheers,
>>>>> >>     >> Daniel
>>>>> >>     >>
>>>>> >>     >> --
>>>>> >>     >> Daniel-Constantin Mierla
>>>>> >>     >> Kamailio (OpenSER) Advanced Training
>>>>> >>     >> Miami, Fl, USA - June 21-23, 2010
>>>>> >>     >> http://www.asipto.com/index.php/kamailio-advanced-training/
>>>>> >>     >>
>>>>> >>     >>
>>>>> >>     >> _______________________________________________
>>>>> >>     >> FreeSWITCH-users mailing list
>>>>> >>     >> FreeSWITCH-users at lists.freeswitch.org
>>>>> >>     >>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >>     >>
>>>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>> >>     >> users
>>>>> >>     >> http://www.freeswitch.org
>>>>> >>     >
>>>>> >>     > _______________________________________________
>>>>> >>     > FreeSWITCH-users mailing list
>>>>> >>     > FreeSWITCH-users at lists.freeswitch.org
>>>>> >>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >>     >
>>>>> >> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-us
>>>>> >>     >ers http://www.freeswitch.org
>>>>> >>
>>>>> >>     --
>>>>> >>     Sincerely,
>>>>> >>
>>>>> >>     Giovanni Maruzzelli
>>>>> >>     Cell : +39-347-2665618
>>>>> >>
>>>>> >>
>>>>> >>     _______________________________________________
>>>>> >>     FreeSWITCH-users mailing list
>>>>> >>     FreeSWITCH-users at lists.freeswitch.org
>>>>> >>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >>
>>>>> >>  UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> >>  http://www.freeswitch.org
>>>>> >>
>>>>> >>
>>>>> >>
>>>>> >>
>>>>> >>
>>>>> >>
>>>>> ---------------------------------------------------------------------------
>>>>> >> ---
>>>>> >>
>>>>> >>
>>>>> >>   _______________________________________________
>>>>> >>   FreeSWITCH-users mailing list
>>>>> >>   FreeSWITCH-users at lists.freeswitch.org
>>>>> >>   http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> >>
>>>>> >> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> >>   http://www.freeswitch.org
>>>>> >>
>>>>> >
>>>>> >
>>>>> > _______________________________________________
>>>>> > FreeSWITCH-users mailing list
>>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> > UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> > http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>  ------------------------------
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>  ------------------------------
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>
>>>
>>> --
>>> Daniel-Constantin Mierla
>>> Kamailio (OpenSER) Advanced Training
>>> Miami, Fl, USA - June 21-23, 2010
>>> http://www.asipto.com/index.php/kamailio-advanced-training/
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From wasim at convergence.pk  Mon May 17 21:19:45 2010
From: wasim at convergence.pk (Wasim Baig)
Date: Tue, 18 May 2010 09:19:45 +0500
Subject: [Freeswitch-users] centos 5.4 timer issue 1000hz,
	choppy voice at times
Message-ID: <AANLkTikhNMygm-3Se-6_xYJ5XgZELe-unaBMzeRMlvQa@mail.gmail.com>

I know its been discussed previously on list, but has there been any
progress to the timer issue on centos 5.4 32 bit

Linux pum2 2.6.18-164.el5 #1 SMP Thu Sep 3 03:33:56 EDT 2009 i686 i686 i386
GNU/Linux

[root at pum2 ~]# grep 1000 /boot/config-2.6.18-164.el5
CONFIG_HZ_1000=y
CONFIG_HZ=1000

i used a basic timer testing utility to confirm what we're actually getting

[root at pum2 ~]# ./timerchk
kernel timer interrupt frequency is approx. 1002 Hz

however, fs complains ...

2010-05-18 06:17:24.711360 [CONSOLE] switch_time.c:1032 Calibrating timer,
please wait...
2010-05-18 06:17:24.861063 [CONSOLE] switch_time.c:223 Test: 1000 Average:
2992 Step: 50
2010-05-18 06:17:24.861097 [WARNING] switch_time.c:227 Abnormally large
timer gap 1992 detected!
Do you have your kernel timer frequency set to lower than 1,000Hz? You may
experience audio problems.

any recommendations, other than drop to 5.3? which is not doable as the box
is a 5k km away ...

-- 
wasim h. baig | principal consultant | convergence pk | +92 30 0850 8070 |
peace be upon you ...
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From babak.freeswitch at gmail.com  Mon May 17 23:25:01 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Tue, 18 May 2010 10:55:01 +0430
Subject: [Freeswitch-users] EventConsumer and multi-threading
In-Reply-To: <AANLkTilKAf1tWJuIrn4WQgQT0lcyX3vIoMNlpejB7fPQ@mail.gmail.com>
References: <AANLkTinz4WHJrO_6q3anFdtiU3wZyaH8XHaaGFjVpQhB@mail.gmail.com>
	<AANLkTilKAf1tWJuIrn4WQgQT0lcyX3vIoMNlpejB7fPQ@mail.gmail.com>
Message-ID: <AANLkTilfC2oKlgkUfBGyo4YRZwtXS8FAByHdb9LkjboF@mail.gmail.com>

thanks
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From babak.freeswitch at gmail.com  Mon May 17 23:29:35 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Tue, 18 May 2010 10:59:35 +0430
Subject: [Freeswitch-users] dll sym error
In-Reply-To: <AANLkTineH1ZIn4Bd-Ll5IXV0q9cVxM7_RuFnGWy6urm-@mail.gmail.com>
References: <AANLkTikfEtaFuN9xiVFDEa11yEnpsM7PgdTQ7h5xFqN-@mail.gmail.com>
	<AANLkTineH1ZIn4Bd-Ll5IXV0q9cVxM7_RuFnGWy6urm-@mail.gmail.com>
Message-ID: <AANLkTikJuJ7Wa77EcKkOehw0_xD0ABMpHN8EAG3ZZBc3@mail.gmail.com>

I was trying to add my module by changing and creating a new project using
mod_say_fr. it was compiling and building but on loading it gave me the
above error. but finally I put it away and changed the mod_say_fr itself and
now everything's fine
thanx for ur concern
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From dome at tel.co.th  Tue May 18 00:44:24 2010
From: dome at tel.co.th (Dome Charoenyost)
Date: Tue, 18 May 2010 14:44:24 +0700
Subject: [Freeswitch-users] Hangup cause when timeout
Message-ID: <AANLkTingRSRk852foYYQxxglRmLqUX4O3ElYjTXZAYsB@mail.gmail.com>

Dear all,
            I want to change hangup cause when bridge time out. now FS
return. 19 	480 	(NO_ANSWER).
I want to change to 34 503 because sender want to do fail over and try
next route. so what's best way to do that ?

modify sofia is my idea :)
BG
Dome C.


From peter.olsson at visionutveckling.se  Tue May 18 00:58:27 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 18 May 2010 09:58:27 +0200
Subject: [Freeswitch-users] centos 5.4 timer issue 1000hz,
 choppy voice at times
In-Reply-To: <AANLkTikhNMygm-3Se-6_xYJ5XgZELe-unaBMzeRMlvQa@mail.gmail.com>
References: <AANLkTikhNMygm-3Se-6_xYJ5XgZELe-unaBMzeRMlvQa@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C557D44DDE1@cooper>

I also have this behaviour (timer gap around 2000). Upgraded yesterday to CentOS 5.5 (from 5.4), but the issue still remains. The system is just used for lab purposes, so I could easily fix SSH access if someone wanted to have a closer look.

I should also add - this is a 3-4 year old machine (workstation), but it seems this problem is related to newer CentOS-kernels, so I guess it shouldn't be the issue here.

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Wasim Baig
Skickat: den 18 maj 2010 06:20
Till: FreeSWITCH Users
?mne: [Freeswitch-users] centos 5.4 timer issue 1000hz, choppy voice at times

I know its been discussed previously on list, but has there been any progress to the timer issue on centos 5.4 32 bit

Linux pum2 2.6.18-164.el5 #1 SMP Thu Sep 3 03:33:56 EDT 2009 i686 i686 i386 GNU/Linux

[root at pum2 ~]# grep 1000 /boot/config-2.6.18-164.el5
CONFIG_HZ_1000=y
CONFIG_HZ=1000

i used a basic timer testing utility to confirm what we're actually getting

[root at pum2 ~]# ./timerchk
kernel timer interrupt frequency is approx. 1002 Hz

however, fs complains ...

2010-05-18 06:17:24.711360 [CONSOLE] switch_time.c:1032 Calibrating timer, please wait...
2010-05-18 06:17:24.861063 [CONSOLE] switch_time.c:223 Test: 1000 Average: 2992 Step: 50
2010-05-18 06:17:24.861097 [WARNING] switch_time.c:227 Abnormally large timer gap 1992 detected!
Do you have your kernel timer frequency set to lower than 1,000Hz? You may experience audio problems.

any recommendations, other than drop to 5.3? which is not doable as the box is a 5k km away ...

--
wasim h. baig | principal consultant | convergence pk | +92 30 0850 8070 | peace be upon you ...
!DSPAM:4bf2176432938985444358!
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From grsingh750 at gmail.com  Tue May 18 02:27:56 2010
From: grsingh750 at gmail.com (guru singh)
Date: Tue, 18 May 2010 14:57:56 +0530
Subject: [Freeswitch-users] Hardware Suggestions.
Message-ID: <AANLkTimDh5tlxUTjskLaNwZiuhnmU-6xLuV7jyXDI1lW@mail.gmail.com>

Hi,

Please recommend a 24 port, full-duplex, PoE switch. I'll be plugging
Cisco 7940 phones into it. Also which gateway router do you recommend?
It needs to have 4 ports and should allow QoS and VLAN tagging. I'd
prefer something that is unlocked and can run linux/openwrt.

Thanks
gs


From david.ponzone at gmail.com  Tue May 18 02:44:02 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Tue, 18 May 2010 11:44:02 +0200
Subject: [Freeswitch-users] Hardware Suggestions.
In-Reply-To: <AANLkTimDh5tlxUTjskLaNwZiuhnmU-6xLuV7jyXDI1lW@mail.gmail.com>
References: <AANLkTimDh5tlxUTjskLaNwZiuhnmU-6xLuV7jyXDI1lW@mail.gmail.com>
Message-ID: <13EAE6CA-2369-4479-BA16-35AE195ABFCB@gmail.com>

Guru,

this would need to be confirmed, but as far as I remember, the regular  
Cisco 7940 POE is not compatible with the standard POE implemented in  
all switches.
You would need to get a Cisco switch, supporting Cisco's non-standard  
POE implementation.

I think this was fixed in the Cisco 7941.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 18/05/2010 ? 11:27, guru singh a ?crit :

> Hi,
>
> Please recommend a 24 port, full-duplex, PoE switch. I'll be plugging
> Cisco 7940 phones into it. Also which gateway router do you recommend?
> It needs to have 4 ports and should allow QoS and VLAN tagging. I'd
> prefer something that is unlocked and can run linux/openwrt.
>
> Thanks
> gs
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jan.berger at video24.no  Tue May 18 05:09:15 2010
From: jan.berger at video24.no (Jan Berger)
Date: Tue, 18 May 2010 14:09:15 +0200
Subject: [Freeswitch-users] SCTP for Windows
Message-ID: <527B81D4113C455A964AAE20C28C3D49@dell9400>

Hi list,

 

Does anyone know about a proper SCTP library for Windows that is reliable
and with a compatible licens?

 

Jan

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From jeremy at seadragons.us  Tue May 18 06:46:32 2010
From: jeremy at seadragons.us (Jeremy Shaffner)
Date: Tue, 18 May 2010 09:46:32 -0400
Subject: [Freeswitch-users] Hardware Suggestions.
In-Reply-To: <AANLkTimDh5tlxUTjskLaNwZiuhnmU-6xLuV7jyXDI1lW@mail.gmail.com>
References: <AANLkTimDh5tlxUTjskLaNwZiuhnmU-6xLuV7jyXDI1lW@mail.gmail.com>
Message-ID: <8A28CB13-215D-41FE-8CCA-02A0E050CC97@seadragons.us>

Hello,

Like David, I would suggest a Cisco switch as well.

For a router, if you want flexibility and Linux, you could use Vyata, either on your own hardware or with one of their appliances.

-Jeremy


On May 18, 2010, at 5:27 AM, guru singh wrote:

> Hi,
> 
> Please recommend a 24 port, full-duplex, PoE switch. I'll be plugging
> Cisco 7940 phones into it. Also which gateway router do you recommend?
> It needs to have 4 ports and should allow QoS and VLAN tagging. I'd
> prefer something that is unlocked and can run linux/openwrt.
> 
> Thanks
> gs
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From freeswitch.org at todandlorna.com  Tue May 18 06:57:38 2010
From: freeswitch.org at todandlorna.com (Tod Hansmann)
Date: Tue, 18 May 2010 07:57:38 -0600
Subject: [Freeswitch-users] SCTP for Windows
In-Reply-To: <527B81D4113C455A964AAE20C28C3D49@dell9400>
References: <527B81D4113C455A964AAE20C28C3D49@dell9400>
Message-ID: <4BF29CD2.4020109@todandlorna.com>

http://www.sctp.org/implementations.html

-Tod Hansmann

On 5/18/2010 6:09 AM, Jan Berger wrote:
>
> Hi list,
>
> Does anyone know about a proper SCTP library for Windows that is 
> reliable and with a compatible licens?
>
> Jan
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    
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From mike at jerris.com  Tue May 18 07:09:05 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 18 May 2010 10:09:05 -0400
Subject: [Freeswitch-users] Fwd: Can you spare 5 minutes and share your
	thoughts on Open Source PBX systems?
References: <4b1644405c284f6380e806e213032ad9@491>
Message-ID: <27B55ABB-0841-4502-99CC-72CBC5CEE6A9@jerris.com>

These are the surveys digium uses to quote usage numbers on open source.  Feel free to fill this out if you like so they can get accurate numbers of the open source PBX marketplace and FreeSWITCH's real footprint.

Mike


http://app.en25.com/e/er.aspx?s=491&lid=680&elq=4b1644405c284f6380e806e213032ad9

Begin forwarded message:

> From: Digium <webannounce at digium.com>
> Date: May 18, 2010 9:58:26 AM EDT
> Subject: Can you spare 5 minutes and share your thoughts on Open Source PBX systems?
> Reply-To: Digium <webannounce at digium.com>
> 
> If you are having trouble reading this email, read the online version.
> 
> 
> 
> 
> Dear Mike,
> 
> Digium? is conducting its third annual survey of organizations and individuals that have installed or may install an Open Source PBX. The survey can be completed in under 5-minutes.
> 
> Everyone completing the survey and providing an email address will receive an entry into our drawing for a NEWiPAD, and highlights of the survey results once we finish the survey.
> 
> Please click the link and complete the survey. We thank you for your participation!
> 
> 
>     Take the Survey    
> Best Regards,
> The Digium Team
> 
> 
> Copyright ? Digium, Inc. - The Asterisk Company
> 445 Jan Davis Drive NW, Huntsville, AL 35806
> Visit our website: Digium.com | Unsubscribe | Update Subscriptions
> 

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From david.ponzone at gmail.com  Tue May 18 07:10:02 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Tue, 18 May 2010 16:10:02 +0200
Subject: [Freeswitch-users] Hardware Suggestions.
In-Reply-To: <8A28CB13-215D-41FE-8CCA-02A0E050CC97@seadragons.us>
References: <AANLkTimDh5tlxUTjskLaNwZiuhnmU-6xLuV7jyXDI1lW@mail.gmail.com>
	<8A28CB13-215D-41FE-8CCA-02A0E050CC97@seadragons.us>
Message-ID: <13753D0D-3002-4A44-B7E8-6DA5F868E48B@gmail.com>

Like Jeremy, I would recommend Vyatta :)
I am not using that in production yet, but I am considering it for my  
BGP core routers.

it's possible to install it in a VM to play with it.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 18/05/2010 ? 15:46, Jeremy Shaffner a ?crit :

> Hello,
>
> Like David, I would suggest a Cisco switch as well.
>
> For a router, if you want flexibility and Linux, you could use  
> Vyata, either on your own hardware or with one of their appliances.
>
> -Jeremy
>
>
> On May 18, 2010, at 5:27 AM, guru singh wrote:
>
>> Hi,
>>
>> Please recommend a 24 port, full-duplex, PoE switch. I'll be plugging
>> Cisco 7940 phones into it. Also which gateway router do you  
>> recommend?
>> It needs to have 4 ports and should allow QoS and VLAN tagging. I'd
>> prefer something that is unlocked and can run linux/openwrt.
>>
>> Thanks
>> gs
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Tue May 18 07:32:49 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 18 May 2010 10:32:49 -0400
Subject: [Freeswitch-users] dll sym error
In-Reply-To: <AANLkTikJuJ7Wa77EcKkOehw0_xD0ABMpHN8EAG3ZZBc3@mail.gmail.com>
References: <AANLkTikfEtaFuN9xiVFDEa11yEnpsM7PgdTQ7h5xFqN-@mail.gmail.com>
	<AANLkTineH1ZIn4Bd-Ll5IXV0q9cVxM7_RuFnGWy6urm-@mail.gmail.com>
	<AANLkTikJuJ7Wa77EcKkOehw0_xD0ABMpHN8EAG3ZZBc3@mail.gmail.com>
Message-ID: <D1F8D34F-8C3F-4137-8924-5A91C163DE6D@jerris.com>

you didn't add the dependencies when you added it to the solution so it was not linked to everything it should be.

Mike

On May 18, 2010, at 2:29 AM, babak yakhchali wrote:

> I was trying to add my module by changing and creating a new project using mod_say_fr. it was compiling and building but on loading it gave me the above error. but finally I put it away and changed the mod_say_fr itself and now everything's fine


From mike at jerris.com  Tue May 18 07:42:05 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 18 May 2010 10:42:05 -0400
Subject: [Freeswitch-users] SCTP for Windows
In-Reply-To: <527B81D4113C455A964AAE20C28C3D49@dell9400>
References: <527B81D4113C455A964AAE20C28C3D49@dell9400>
Message-ID: <B80AF699-A323-4031-8900-3AD3081912CB@jerris.com>

http://www.co-conv.jp/en/product/sctpDrv/20081224/

On May 18, 2010, at 8:09 AM, Jan Berger wrote:

> Hi list,
>  
> Does anyone know about a proper SCTP library for Windows that is reliable and with a compatible licens?

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From anthony.minessale at gmail.com  Tue May 18 08:35:06 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 18 May 2010 10:35:06 -0500
Subject: [Freeswitch-users] centos 5.4 timer issue 1000hz,
	choppy voice at 	times
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C557D44DDE1@cooper>
References: <AANLkTikhNMygm-3Se-6_xYJ5XgZELe-unaBMzeRMlvQa@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C557D44DDE1@cooper>
Message-ID: <AANLkTinAK8jYwaSdr4hcmv5j15Wj7HxHe5HqCV4x6Ss8@mail.gmail.com>

Please heed as this is our official policy.

There is not an issue with CentOS 5.3.
In fact CentOS 5.3 is the preferred platform (not 5.4 or 5.5 until we do
proper testing and QA).

What you are describing is the result of poor hardware, (most likely the
motherboard)

FreeSWITCH is Free, your box is not, use the money you save not buying
FreeSWITCH to buy a decent box.  Even a $500 box can work perfectly fine.
 Its not my intent to cater to inferior hardware when my goal is to push
telephony into the future not get caught up in legacy drama with older
boxes.

If you have altered the kernel to run at something besides 1000hz, change it
back to the default of 1000hz.   If you get the error instantly it means the
kernel is actually configured at the wrong freq.
If the test runs for a long time and concludes it, it most likely means the
box is not good enough to run FS.

If you are not careful you may have installed the xen build of the kernel
(check with uname -a)
This can also cause timing issues.

If you see that timing gap warning (especially on centos if you have it set
to 1000hz) it means the motherboard cannot give the OS a good enough timing
source to provide 1ms resolution which is mandatory and not very demanding.

I have been on dozens of perfectly happy boxes running CentOS 5.3 and I have
been on a small handful of boxes that do what you describe and have seen
that, once they are where they belong, (in the trash) the new inexpensive
replacement does just fine.


On Tue, May 18, 2010 at 2:58 AM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

>  I also have this behaviour (timer gap around 2000). Upgraded yesterday to
> CentOS 5.5 (from 5.4), but the issue still remains. The system is just used
> for lab purposes, so I could easily fix SSH access if someone wanted to have
> a closer look.
>
>
>
> I should also add ? this is a 3-4 year old machine (workstation), but it
> seems this problem is related to newer CentOS-kernels, so I guess it
> shouldn?t be the issue here.
>
>
>
> /Peter
>
>
>
> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *F?r *Wasim Baig
> *Skickat:* den 18 maj 2010 06:20
> *Till:* FreeSWITCH Users
> *?mne:* [Freeswitch-users] centos 5.4 timer issue 1000hz, choppy voice at
> times
>
>
>
> I know its been discussed previously on list, but has there been any
> progress to the timer issue on centos 5.4 32 bit
>
>
>
> Linux pum2 2.6.18-164.el5 #1 SMP Thu Sep 3 03:33:56 EDT 2009 i686 i686 i386
> GNU/Linux
>
>
>
> [root at pum2 ~]# grep 1000 /boot/config-2.6.18-164.el5
>
> CONFIG_HZ_1000=y
>
> CONFIG_HZ=1000
>
>
>
> i used a basic timer testing utility to confirm what we're actually getting
>
>
>
> [root at pum2 ~]# ./timerchk
>
> kernel timer interrupt frequency is approx. 1002 Hz
>
>
>
> however, fs complains ...
>
>
>
> 2010-05-18 06:17:24.711360 [CONSOLE] switch_time.c:1032 Calibrating timer,
> please wait...
>
> 2010-05-18 06:17:24.861063 [CONSOLE] switch_time.c:223 Test: 1000 Average:
> 2992 Step: 50
>
> 2010-05-18 06:17:24.861097 [WARNING] switch_time.c:227 Abnormally large
> timer gap 1992 detected!
>
> Do you have your kernel timer frequency set to lower than 1,000Hz? You may
> experience audio problems.
>
>
>
> any recommendations, other than drop to 5.3? which is not doable as the box
> is a 5k km away ...
>
>
>
> --
> wasim h. baig | principal consultant | convergence pk | +92 30 0850 8070 |
> peace be upon you ...
>
> !DSPAM:4bf2176432938985444358!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <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
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From kenfulmer at icstechnologysolutions.com  Tue May 18 08:40:20 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Tue, 18 May 2010 10:40:20 -0500
Subject: [Freeswitch-users] Contact Field
Message-ID: <008401caf6a0$6c87a670$4596f350$@com>

We are connecting a FS gateway to PaeTec. They need the contact field to be
the same as the calling number. Using the trunk registration 10 digit number
doesn't work for them.

 

I added the following parameter to remove "gw + external" in the contact
field to the left of the @ sign:

 

    <param name="extension-in-contact" value="true"/>

 

Is there something else I should add, so the contact field shows the 10
digit calling number? 

 

Thanks,

 

Ken Fulmer

 

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From lloyd.aloysius at sunteltech.ca  Tue May 18 08:56:28 2010
From: lloyd.aloysius at sunteltech.ca (Aloysius Lloyd)
Date: Tue, 18 May 2010 11:56:28 -0400
Subject: [Freeswitch-users] Contact Field
In-Reply-To: <008401caf6a0$6c87a670$4596f350$@com>
References: <008401caf6a0$6c87a670$4596f350$@com>
Message-ID: <AANLkTimFbMQWeeGkcO6Mlax85suWi2d7GH_glD8K5usn@mail.gmail.com>

HI Ken,

Add the following line

<param name="extension" value="10 Digit Number"/>

Lloyd


On Tue, May 18, 2010 at 11:40 AM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  We are connecting a FS gateway to PaeTec. They need the contact field to
> be the same as the calling number. Using the trunk registration 10 digit
> number doesn?t work for them.
>
>
>
> I added the following parameter to remove ?gw + external? in the contact
> field to the left of the @ sign:
>
>
>
>     <param name="extension-in-contact" value="true"/>
>
>
>
> Is there something else I should add, so the contact field shows the 10
> digit calling number?
>
>
>
> Thanks,
>
>
>
> Ken Fulmer
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From neilp at cs.stanford.edu  Tue May 18 09:07:14 2010
From: neilp at cs.stanford.edu (Neil Patel)
Date: Tue, 18 May 2010 09:07:14 -0700
Subject: [Freeswitch-users] span not defined error
In-Reply-To: <AANLkTim6OwUdrXQsdsSo9npyMVaezHlVLfhZfNGszEQP@mail.gmail.com>
References: <AANLkTik9DQcT5fpgcfb6KqFUZT4vXUuAjqTE5pFLxW-R@mail.gmail.com> 
	<AANLkTin9wP0f74qEyA5GjtpVo7YacbS2C6VE9VnO31wj@mail.gmail.com> 
	<AANLkTikleSqsMITY0lNf-Qyeg5kOb2ICJFDs4gO3UeZO@mail.gmail.com> 
	<AANLkTikd_xI3QFUeWjJfVzNs5nc1tuK9_WHEu46Jsekd@mail.gmail.com> 
	<AANLkTinbeTgmLDAcxyF3B_JqO7J59XFTbTOtLio0ByaM@mail.gmail.com> 
	<AANLkTim6OwUdrXQsdsSo9npyMVaezHlVLfhZfNGszEQP@mail.gmail.com>
Message-ID: <AANLkTikNAel0cKvf0h-8vO-bdhXU_WO_35kiWWWSC7BS@mail.gmail.com>

Hey Moises,

Upgrading wanpipe and FS to the latest did the trick. Thanks!

-Neil

On Mon, May 17, 2010 at 7:44 AM, Moises Silva <moises.silva at gmail.com>wrote:

> Try taking a protocol dump using wanpipemon on the d-channel so we can see
> the full details of the call setup.
>
>
> http://wiki.sangoma.com/wanpipe-wireshark-pcap-pri-bri-wan-t1-e1-tracing#pri_bri_wireshark
>
> Then paste a link here to the pcap file somewhere.
>
> Moises Silva
> Senior Software Engineer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
> 9T3 Canada
> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>
>
> On Sun, May 16, 2010 at 8:16 PM, Neil Patel <neilp at cs.stanford.edu> wrote:
>
>> Thanks Moises, this is very helpful!
>>
>> After fixing the dialstring, I am still getting the following error from
>> FS:
>>
>> > originate openzap/smg_prid/a/<num> &echo
>> -ERR NORMAL_CIRCUIT_CONGESTION
>>
>> 2010-05-17 05:39:02.883264 [WARNING] ozmod_sangoma_boost.c:346 TX EVENT:
>> CALL_START:(80) [w1g1] CSid=1 Seq=0 Cn=[FreeSWITCH] Cd=[9428500597]
>> Ci=[0000000000] Rdnis=[]
>> freeswitch at internal> 2010-05-17 05:39:02.955336 [WARNING]
>> ozmod_sangoma_boost.c:1312 RX EVENT (N): CALL_START_NACK:(82) [w256g256]
>> Rc=0 CSid=1 Seq=1
>> 2010-05-17 05:39:02.955336 [WARNING] sangoma_boost_client.c:220 TX EVENT
>> (N): CALL_START_NACK_ACK:(83) [w1g1] Rc=0 CSid=1 Seq=1
>> 2010-05-17 05:39:02.956586 [ERR] mod_openzap.c:1154 No channels available
>> 2010-05-17 05:39:02.956586 [ERR] switch_ivr_originate.c:2249 Cannot create
>> outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
>>
>>
>>
>> Here is the relevant logging from /var/log/sangoma_mgd.log:
>>
>> May 17 05:39:02 otalo sangoma_prid: Rx SMG [CALL_START s:0 c:0 id:1]
>> May 17 05:39:02 otalo sangoma_prid: g1:Outgoing call (Smg-ID:1)
>> May 17 05:39:02 otalo sangoma_prid: s1:Outgoing call ChanRq:1
>> Called-Nb[9428500597] Calling-Nb[0000000000] (Smg-ID:1)
>> May 17 05:39:02 otalo sangoma_prid: Rx Tsoft [7:StatusIn id:0]
>> May 17 05:39:02 otalo sangoma_prid: s1:Received cause-ind-An IE or
>> parameter does not exist(99)
>> May 17 05:39:02 otalo sangoma_prid: Rx Tsoft [2:DiscIn id:0]
>> May 17 05:39:02 otalo sangoma_prid: s1c1:Remote released-Unknown(0)
>> May 17 05:39:02 otalo sangoma_prid: Rx SMG [CALL_START_NACK_ACK s:0 c:0
>> id:1]
>> May 17 05:39:02 otalo sangoma_prid: g1:Call cleared (SMG-ID:1)
>>
>>
>> I updated /etc/wanpipe/smg_pri.conf to have group=1 uncommented.
>>
>> Please note that my box is in India, making calls over a PRI line set up
>> there. I also changed my tones.conf to match what's on the wiki<http://wiki.freeswitch.org/wiki/Tones.conf_Example>for India.
>>
>> Thanks,
>> Neil
>>
>> On Sat, May 15, 2010 at 10:20 PM, Moises Silva <moises.silva at gmail.com>wrote:
>>
>>> Hi Again Neil,
>>>
>>> I just noticed your dial string is incorrect. The correct syntax is:
>>>
>>> OpenZAP/<span>/<chan_code>/[number]
>>>
>>> The span and chan code are mandatory. The number is optional ( FXS
>>> channels do not require a number, they just ring the FXO device connected to
>>> them).
>>>
>>> The span is either a number ( span id, the id is a number assigned in the
>>> order in which the span is defined in openzap.conf ) or a span name also as
>>> specified in the [span wanpipe <span_name>] line in openzap.conf
>>>
>>> The chan code is either a number ( for spans that support individual
>>> channel selection, boost is NOT one of them ), or a channel hunting mode,
>>> there is currently 2 modes, "a" is top down and "A" is bottom up.
>>>
>>> So, this is a valid string for you case:
>>>
>>> OpenZAP/smg_prid/a/<number>
>>>
>>> In the specific case of boost in socket mode ( openzap only supports
>>> socket mode ) the number may contain @gX where X is a group ( for hunting as
>>> configured in /etc/wanpipe/smg_pri.conf). Boost signaling are a special case
>>> because the hunting for channels is not done within FreeSWITCH but in
>>> sangoma_prid binary ( in the new OpenZAP version called FreeTDM this has
>>> changed depending on configuration).
>>>
>>> Bottom line, this should work:
>>>
>>> OpenZAP/smg_prid/a/1234 at g1
>>>
>>> If you have g1 configured in /etc/wanpipe/smg_pri.conf
>>>
>>> Moises Silva
>>> Senior Software Engineer
>>> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
>>> 9T3 Canada
>>> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>>>
>>>
>>> On Sat, May 15, 2010 at 6:26 PM, Neil Patel <neilp at cs.stanford.edu>wrote:
>>>
>>>> Span was originally in the boost section when I got this error, so I
>>>> thought I'd try it in analog and both. None work.
>>>>
>>>> -Neil
>>>>
>>>> On Sat, May 15, 2010 at 3:18 PM, Moises Silva <moises.silva at gmail.com>wrote:
>>>>
>>>>> Why do you have 2 spans in openzap.conf.xml with the same name, in both
>>>>> the boost and analog sections?
>>>>> Moises Silva
>>>>> Senior Software Engineer
>>>>> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON
>>>>> L3R 9T3 Canada
>>>>> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>>>>>
>>>>>
>>>>> On Sat, May 15, 2010 at 3:52 PM, Neil Patel <neilp at cs.stanford.edu>wrote:
>>>>>
>>>>>> Hi All,
>>>>>>
>>>>>> I am trying to dial out over my PRI, and am getting this error:
>>>>>>
>>>>>> 2010-05-16 01:01:21.452392 [CRIT] zap_io.c:1139 SPAN NOT DEFINED!
>>>>>> 2010-05-16 01:01:21.452392 [ERR] mod_openzap.c:1154 No channels
>>>>>> available
>>>>>> 2010-05-16 01:01:21.452392 [ERR] switch_ivr_originate.c:2249 Cannot
>>>>>> create outgoing channel of type [OpenZAP] cause: [NORMAL_CIRCUIT_CONGESTION]
>>>>>>
>>>>>>
>>>>>> This is my openzap.conf:
>>>>>>
>>>>>> [span wanpipe smg_prid]
>>>>>> name => smg_prid
>>>>>> trunk_type =>e1
>>>>>> b-channel => 1:1-15
>>>>>> b-channel => 1:17-31
>>>>>> trunk_type =>e1
>>>>>> b-channel => 2:1-15
>>>>>> b-channel => 2:17-31
>>>>>>
>>>>>>
>>>>>> This is my openzap.conf.xml:
>>>>>>
>>>>>> <configuration name="openzap.conf" description="OpenZAP
>>>>>> Configuration">
>>>>>>   <settings>
>>>>>>     <param name="debug" value="0"/>
>>>>>>
>>>>>>   </settings>
>>>>>>   <boost_spans>
>>>>>>         <span id="smg_prid">
>>>>>>         <param name="dialplan" value="XML"/>
>>>>>>         <param name="context" value="default"/>
>>>>>>         </span>
>>>>>>   </boost_spans>
>>>>>>
>>>>>>   <analog_spans>
>>>>>>         <span id="smg_prid">
>>>>>>         <param name="dialplan" value="XML"/>
>>>>>>         <param name="context" value="default"/>
>>>>>>
>>>>>>         </span>
>>>>>>   </analog_spans>
>>>>>> </configuration>
>>>>>>
>>>>>>
>>>>>> And here is the lua code I'm using to dial out:
>>>>>>
>>>>>> sessiondata = "OpenZAP/smg_prid/<num>"
>>>>>> new_session = freeswitch.Session(sessiondata)
>>>>>>
>>>>>>
>>>>>> What am I missing here?
>>>>>> Thanks,
>>>>>> Neil
>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From grsingh750 at gmail.com  Tue May 18 09:14:19 2010
From: grsingh750 at gmail.com (guru singh)
Date: Tue, 18 May 2010 21:44:19 +0530
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 47, Issue 164
In-Reply-To: <mailman.3577.1274196940.19150.freeswitch-users@lists.freeswitch.org>
References: <mailman.3577.1274196940.19150.freeswitch-users@lists.freeswitch.org>
Message-ID: <AANLkTilr4J_r_woeGU4ZN_owdcAEM_P957Rn4Hd6Dxpo@mail.gmail.com>

> ---------- Forwarded message ----------
> From:?David Ponzone <david.ponzone at gmail.com>
> To:?freeswitch-users at lists.freeswitch.org
> Date:?Tue, 18 May 2010 16:10:02 +0200
> Subject:?Re: [Freeswitch-users] Hardware Suggestions.
> Like Jeremy, I would recommend Vyatta :)
> I am not using that in production yet, but I am considering it for my BGP core routers.
> it's possible to install it in a VM to play with it.

Thanks, Vyatta looks just appropriate.
Also, I think you're right about the Cisco 7940 phones not working
with standard PoE appliances.
Since, I'm getting refurbished models at a very moderate price, I
think the trade-off in buying a Cisco PoE switch is worth it?

thanks
gs

> David Ponzone ?Direction Technique
> email: david.ponzone at ipeva.fr
> tel: ? ? ?01 74 03 18 97
> gsm: ? 06 66 98 76 34
> Service Client?IPeva
> tel: ? ? ?0811 46 26 26


From peter.olsson at visionutveckling.se  Tue May 18 09:24:59 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 18 May 2010 18:24:59 +0200
Subject: [Freeswitch-users] centos 5.4 timer issue 1000hz,
 choppy voice at 	times
In-Reply-To: <AANLkTinAK8jYwaSdr4hcmv5j15Wj7HxHe5HqCV4x6Ss8@mail.gmail.com>
References: <AANLkTikhNMygm-3Se-6_xYJ5XgZELe-unaBMzeRMlvQa@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C557D44DDE1@cooper>
	<AANLkTinAK8jYwaSdr4hcmv5j15Wj7HxHe5HqCV4x6Ss8@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C557D44E082@cooper>

Anthony,

Just so you didn't get me wrong (feels like I got stuck in the middle here...). I Just wanted to point out that I do have the same "problem" as Wasim. I'm well aware I'm running this on old hardware, I just use it for lab purposes, so it really doesn't matter. I don't have any audio problems, since I just use it for 1-2 silutaneous calls. But I do get the warning when I start up FS, the same on both CentOS 5.4 and 5.5 (didn't try 5.3 on this box, I started with 5.4 and upgraded yesterday), so I just wanted to try helping out if anyone needed a 5.5 box to try on.

I'm running "production" (only for internal use, but still) servers on both CentOS 5.3 and Windows 2003, which work perfectly fine (no timer gap warnings) - but they have good hardware though :)

Regards,
Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anthony Minessale
Skickat: den 18 maj 2010 17:35
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] centos 5.4 timer issue 1000hz, choppy voice at times

Please heed as this is our official policy.

There is not an issue with CentOS 5.3.
In fact CentOS 5.3 is the preferred platform (not 5.4 or 5.5 until we do proper testing and QA).

What you are describing is the result of poor hardware, (most likely the motherboard)

FreeSWITCH is Free, your box is not, use the money you save not buying FreeSWITCH to buy a decent box.  Even a $500 box can work perfectly fine.  Its not my intent to cater to inferior hardware when my goal is to push telephony into the future not get caught up in legacy drama with older boxes.

If you have altered the kernel to run at something besides 1000hz, change it back to the default of 1000hz.   If you get the error instantly it means the kernel is actually configured at the wrong freq.
If the test runs for a long time and concludes it, it most likely means the box is not good enough to run FS.

If you are not careful you may have installed the xen build of the kernel (check with uname -a)
This can also cause timing issues.

If you see that timing gap warning (especially on centos if you have it set to 1000hz) it means the motherboard cannot give the OS a good enough timing source to provide 1ms resolution which is mandatory and not very demanding.

I have been on dozens of perfectly happy boxes running CentOS 5.3 and I have been on a small handful of boxes that do what you describe and have seen that, once they are where they belong, (in the trash) the new inexpensive replacement does just fine.


On Tue, May 18, 2010 at 2:58 AM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:
I also have this behaviour (timer gap around 2000). Upgraded yesterday to CentOS 5.5 (from 5.4), but the issue still remains. The system is just used for lab purposes, so I could easily fix SSH access if someone wanted to have a closer look.

I should also add - this is a 3-4 year old machine (workstation), but it seems this problem is related to newer CentOS-kernels, so I guess it shouldn't be the issue here.

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] F?r Wasim Baig
Skickat: den 18 maj 2010 06:20
Till: FreeSWITCH Users
?mne: [Freeswitch-users] centos 5.4 timer issue 1000hz, choppy voice at times

I know its been discussed previously on list, but has there been any progress to the timer issue on centos 5.4 32 bit

Linux pum2 2.6.18-164.el5 #1 SMP Thu Sep 3 03:33:56 EDT 2009 i686 i686 i386 GNU/Linux

[root at pum2 ~]# grep 1000 /boot/config-2.6.18-164.el5
CONFIG_HZ_1000=y
CONFIG_HZ=1000

i used a basic timer testing utility to confirm what we're actually getting

[root at pum2 ~]# ./timerchk
kernel timer interrupt frequency is approx. 1002 Hz

however, fs complains ...

2010-05-18 06:17:24.711360 [CONSOLE] switch_time.c:1032 Calibrating timer, please wait...
2010-05-18 06:17:24.861063 [CONSOLE] switch_time.c:223 Test: 1000 Average: 2992 Step: 50
2010-05-18 06:17:24.861097 [WARNING] switch_time.c:227 Abnormally large timer gap 1992 detected!
Do you have your kernel timer frequency set to lower than 1,000Hz? You may experience audio problems.

any recommendations, other than drop to 5.3? which is not doable as the box is a 5k km away ...

--
wasim h. baig | principal consultant | convergence pk | +92 30 0850 8070 | peace be upon you ...

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



--
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>
googletalk:conf+888 at conference.freeswitch.org<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From kris at kriskinc.com  Tue May 18 09:29:38 2010
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Tue, 18 May 2010 12:29:38 -0400
Subject: [Freeswitch-users] Contact Field
Message-ID: <0b56b97ea4e1c52f05eea74c215aeaf1@mail.gmail.com>

 I assume he is talking about the contact field in the INVITE, not the
REGISTER...

I don't know how to accomplish this off the top of my head but I'd be
interested too (quite a few carriers request this).


--
Kristian Kielhofner
http://blog.krisk.org

------------------------------
 *From*: freeswitch-users-bounces at lists.freeswitch.org <
freeswitch-users-bounces at lists.freeswitch.org>
*To*: freeswitch-users at lists.freeswitch.org <
freeswitch-users at lists.freeswitch.org>
*Sent*: Tue May 18 11:56:28 2010
*Subject*: Re: [Freeswitch-users] Contact Field

HI Ken,

Add the following line

<param name="extension" value="10 Digit Number"/>

Lloyd


On Tue, May 18, 2010 at 11:40 AM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  We are connecting a FS gateway to PaeTec. They need the contact field to
> be the same as the calling number. Using the trunk registration 10 digit
> number doesn?t work for them.
>
>
>
> I added the following parameter to remove ?gw + external? in the contact
> field to the left of the @ sign:
>
>
>
>     <param name="extension-in-contact" value="true"/>
>
>
>
> Is there something else I should add, so the contact field shows the 10
> digit calling number?
>
>
>
> Thanks,
>
>
>
> Ken Fulmer
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jan.berger at video24.no  Tue May 18 09:42:35 2010
From: jan.berger at video24.no (Jan Berger)
Date: Tue, 18 May 2010 18:42:35 +0200
Subject: [Freeswitch-users] SCTP for Windows
In-Reply-To: <B80AF699-A323-4031-8900-3AD3081912CB@jerris.com>
References: <527B81D4113C455A964AAE20C28C3D49@dell9400>
	<B80AF699-A323-4031-8900-3AD3081912CB@jerris.com>
Message-ID: <20100518164235.3F93FD6247B00@bmail02.one.com>


Thanks mate,
?
Japanese ? Might actually work then :)
?
There is a German library as well, but it says GPL.
?
Jan?
On May 18, 2010 16:42 "Michael Jerris" <mike at jerris.com>wrote:
> <http://www.co-conv.jp/en/product/sctpDrv/20081224/>
> On May 18, 2010, at 8:09 AM, Jan Berger wrote:
> > Hi list,
> > ?
> > Does anyone know about a proper SCTP library for Windows that is
> > reliable and with a compatible licens?
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From anthony.minessale at gmail.com  Tue May 18 09:48:55 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 18 May 2010 11:48:55 -0500
Subject: [Freeswitch-users] centos 5.4 timer issue 1000hz,
	choppy voice at 	times
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C557D44E082@cooper>
References: <AANLkTikhNMygm-3Se-6_xYJ5XgZELe-unaBMzeRMlvQa@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C557D44DDE1@cooper>
	<AANLkTinAK8jYwaSdr4hcmv5j15Wj7HxHe5HqCV4x6Ss8@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C557D44E082@cooper>
Message-ID: <AANLkTinMsQSF4hVso46UZ8gw3cqo9NusNClPgpXhIWOM@mail.gmail.com>

Don't worry I am just trying to post our policy so I don't have to repeat
it.
I don't get mad at stuff like this, i'm just being assertive.


On Tue, May 18, 2010 at 11:24 AM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

>  Anthony,
>
>
>
> Just so you didn?t get me wrong (feels like I got stuck in the middle
> here...). I Just wanted to point out that I do have the same ?problem? as
> Wasim. I?m well aware I?m running this on old hardware, I just use it for
> lab purposes, so it really doesn?t matter. I don?t have any audio problems,
> since I just use it for 1-2 silutaneous calls. But I do get the warning when
> I start up FS, the same on both CentOS 5.4 and 5.5 (didn?t try 5.3 on this
> box, I started with 5.4 and upgraded yesterday), so I just wanted to try
> helping out if anyone needed a 5.5 box to try on.
>
>
>
> I?m running ?production? (only for internal use, but still) servers on both
> CentOS 5.3 and Windows 2003, which work perfectly fine (no timer gap
> warnings) ? but they have good hardware though :)
>
>
>
> Regards,
>
> Peter
>
>
>
> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *F?r *Anthony Minessale
> *Skickat:* den 18 maj 2010 17:35
> *Till:* freeswitch-users at lists.freeswitch.org
> *?mne:* Re: [Freeswitch-users] centos 5.4 timer issue 1000hz, choppy voice
> at times
>
>
>
> Please heed as this is our official policy.
>
>
>
> There is not an issue with CentOS 5.3.
>
> In fact CentOS 5.3 is the preferred platform (not 5.4 or 5.5 until we do
> proper testing and QA).
>
>
>
> What you are describing is the result of poor hardware, (most likely the
> motherboard)
>
>
>
> FreeSWITCH is Free, your box is not, use the money you save not buying
> FreeSWITCH to buy a decent box.  Even a $500 box can work perfectly fine.
>  Its not my intent to cater to inferior hardware when my goal is to push
> telephony into the future not get caught up in legacy drama with older
> boxes.
>
>
>
> If you have altered the kernel to run at something besides 1000hz, change
> it back to the default of 1000hz.   If you get the error instantly it means
> the kernel is actually configured at the wrong freq.
>
> If the test runs for a long time and concludes it, it most likely means the
> box is not good enough to run FS.
>
>
>
> If you are not careful you may have installed the xen build of the kernel
> (check with uname -a)
>
> This can also cause timing issues.
>
>
>
> If you see that timing gap warning (especially on centos if you have it set
> to 1000hz) it means the motherboard cannot give the OS a good enough timing
> source to provide 1ms resolution which is mandatory and not very demanding.
>
>
>
> I have been on dozens of perfectly happy boxes running CentOS 5.3 and I
> have been on a small handful of boxes that do what you describe and have
> seen that, once they are where they belong, (in the trash) the new
> inexpensive replacement does just fine.
>
>
>
>
>
> On Tue, May 18, 2010 at 2:58 AM, Peter Olsson <
> peter.olsson at visionutveckling.se> wrote:
>
> I also have this behaviour (timer gap around 2000). Upgraded yesterday to
> CentOS 5.5 (from 5.4), but the issue still remains. The system is just used
> for lab purposes, so I could easily fix SSH access if someone wanted to have
> a closer look.
>
>
>
> I should also add ? this is a 3-4 year old machine (workstation), but it
> seems this problem is related to newer CentOS-kernels, so I guess it
> shouldn?t be the issue here.
>
>
>
> /Peter
>
>
>
> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *F?r *Wasim Baig
> *Skickat:* den 18 maj 2010 06:20
> *Till:* FreeSWITCH Users
> *?mne:* [Freeswitch-users] centos 5.4 timer issue 1000hz, choppy voice at
> times
>
>
>
> I know its been discussed previously on list, but has there been any
> progress to the timer issue on centos 5.4 32 bit
>
>
>
> Linux pum2 2.6.18-164.el5 #1 SMP Thu Sep 3 03:33:56 EDT 2009 i686 i686 i386
> GNU/Linux
>
>
>
> [root at pum2 ~]# grep 1000 /boot/config-2.6.18-164.el5
>
> CONFIG_HZ_1000=y
>
> CONFIG_HZ=1000
>
>
>
> i used a basic timer testing utility to confirm what we're actually getting
>
>
>
> [root at pum2 ~]# ./timerchk
>
> kernel timer interrupt frequency is approx. 1002 Hz
>
>
>
> however, fs complains ...
>
>
>
> 2010-05-18 06:17:24.711360 [CONSOLE] switch_time.c:1032 Calibrating timer,
> please wait...
>
> 2010-05-18 06:17:24.861063 [CONSOLE] switch_time.c:223 Test: 1000 Average:
> 2992 Step: 50
>
> 2010-05-18 06:17:24.861097 [WARNING] switch_time.c:227 Abnormally large
> timer gap 1992 detected!
>
> Do you have your kernel timer frequency set to lower than 1,000Hz? You may
> experience audio problems.
>
>
>
> any recommendations, other than drop to 5.3? which is not doable as the box
> is a 5k km away ...
>
>
>
> --
> wasim h. baig | principal consultant | convergence pk | +92 30 0850 8070 |
> peace be upon you ...
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> 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
>
> !DSPAM:4bf2b5fb32931992811684!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <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
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From david.ponzone at gmail.com  Tue May 18 10:01:29 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Tue, 18 May 2010 19:01:29 +0200
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 47, Issue 164
In-Reply-To: <AANLkTilr4J_r_woeGU4ZN_owdcAEM_P957Rn4Hd6Dxpo@mail.gmail.com>
References: <mailman.3577.1274196940.19150.freeswitch-users@lists.freeswitch.org>
	<AANLkTilr4J_r_woeGU4ZN_owdcAEM_P957Rn4Hd6Dxpo@mail.gmail.com>
Message-ID: <90F3F859-8DD1-4289-90F5-F9AC7C0EA805@gmail.com>

Agreed.
And you may find a refurbished Cisco POE switch :)

For the router part, you may also check Zeroshell (http://www.zeroshell.net/eng/ 
).
It's more CPE-oriented and more multi-purpose than Vyatta.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 18/05/2010 ? 18:14, guru singh a ?crit :

>> ---------- Forwarded message ----------
>> From: David Ponzone <david.ponzone at gmail.com>
>> To: freeswitch-users at lists.freeswitch.org
>> Date: Tue, 18 May 2010 16:10:02 +0200
>> Subject: Re: [Freeswitch-users] Hardware Suggestions.
>> Like Jeremy, I would recommend Vyatta :)
>> I am not using that in production yet, but I am considering it for  
>> my BGP core routers.
>> it's possible to install it in a VM to play with it.
>
> Thanks, Vyatta looks just appropriate.
> Also, I think you're right about the Cisco 7940 phones not working
> with standard PoE appliances.
> Since, I'm getting refurbished models at a very moderate price, I
> think the trade-off in buying a Cisco PoE switch is worth it?
>
> thanks
> gs
>
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> Service Client IPeva
>> tel:      0811 46 26 26
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Tue May 18 10:05:43 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 18 May 2010 12:05:43 -0500
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 47, Issue 164
In-Reply-To: <90F3F859-8DD1-4289-90F5-F9AC7C0EA805@gmail.com>
References: <mailman.3577.1274196940.19150.freeswitch-users@lists.freeswitch.org>
	<AANLkTilr4J_r_woeGU4ZN_owdcAEM_P957Rn4Hd6Dxpo@mail.gmail.com>
	<90F3F859-8DD1-4289-90F5-F9AC7C0EA805@gmail.com>
Message-ID: <E9BF2A3C-11EE-4CBB-8528-A4ACD91A53FA@freeswitch.org>

If anyone is going to respond to list posts please do not subscribe to it with Digest... because nobody can follow it.

/b

On May 18, 2010, at 12:01 PM, David Ponzone wrote:

> Agreed.
> And you may find a refurbished Cisco POE switch :)
> 
> For the router part, you may also check Zeroshell (http://www.zeroshell.net/eng/).
> It's more CPE-oriented and more multi-purpose than Vyatta.
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 

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From david.ponzone at gmail.com  Tue May 18 10:26:23 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Tue, 18 May 2010 19:26:23 +0200
Subject: [Freeswitch-users] Hardware Suggestions.
In-Reply-To: <13753D0D-3002-4A44-B7E8-6DA5F868E48B@gmail.com>
References: <AANLkTimDh5tlxUTjskLaNwZiuhnmU-6xLuV7jyXDI1lW@mail.gmail.com>
	<8A28CB13-215D-41FE-8CCA-02A0E050CC97@seadragons.us>
	<13753D0D-3002-4A44-B7E8-6DA5F868E48B@gmail.com>
Message-ID: <3ED44D68-2A5A-4FD3-B648-55A98FF5FEC2@gmail.com>

(sorry for this repost, but the previous one was by mistake sent with  
the wrong subject, which was bad for the readability of this thread).

Guru,

you may find a refurbished Cisco "POE" switch :)

For the router part, you may also check Zeroshell (http://www.zeroshell.net/eng/ 
).
It's more CPE-oriented and more multi-purpose than Vyatta.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 18/05/2010 ? 16:10, David Ponzone a ?crit :

> Like Jeremy, I would recommend Vyatta :)
> I am not using that in production yet, but I am considering it for  
> my BGP core routers.
>
> it's possible to install it in a VM to play with it.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et  
> ?tablis ? l'intention exclusive de ses destinataires. Toute  
> utilisation ou diffusion non autoris?e est interdite. Tout message  
> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
> responsabilit? au titre de ce message s'il a ?t? alt?r?,  
> d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce  
> message, merci de le d?truire imm?diatement et d'avertir  
> l'exp?diteur.
>
>
>
>
> Le 18/05/2010 ? 15:46, Jeremy Shaffner a ?crit :
>
>> Hello,
>>
>> Like David, I would suggest a Cisco switch as well.
>>
>> For a router, if you want flexibility and Linux, you could use  
>> Vyata, either on your own hardware or with one of their appliances.
>>
>> -Jeremy
>>
>>
>> On May 18, 2010, at 5:27 AM, guru singh wrote:
>>
>>> Hi,
>>>
>>> Please recommend a 24 port, full-duplex, PoE switch. I'll be  
>>> plugging
>>> Cisco 7940 phones into it. Also which gateway router do you  
>>> recommend?
>>> It needs to have 4 ports and should allow QoS and VLAN tagging. I'd
>>> prefer something that is unlocked and can run linux/openwrt.
>>>
>>> Thanks
>>> gs
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>

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From kenfulmer at icstechnologysolutions.com  Tue May 18 10:50:02 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Tue, 18 May 2010 12:50:02 -0500
Subject: [Freeswitch-users] Contact Field
In-Reply-To: <0b56b97ea4e1c52f05eea74c215aeaf1@mail.gmail.com>
References: <0b56b97ea4e1c52f05eea74c215aeaf1@mail.gmail.com>
Message-ID: <00cb01caf6b2$8ae3b1b0$a0ab1510$@com>

Yes, I'm asking about the contact field in the INVITE header. PaeTec won't
accept our calls without this change.

 

Thanks,

 

Ken

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kristian
Kielhofner
Sent: Tuesday, May 18, 2010 11:30 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Contact Field

 

I assume he is talking about the contact field in the INVITE, not the
REGISTER...

I don't know how to accomplish this off the top of my head but I'd be
interested too (quite a few carriers request this).


-- 
Kristian Kielhofner 
http://blog.krisk.org

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
<freeswitch-users-bounces at lists.freeswitch.org> 
To: freeswitch-users at lists.freeswitch.org
<freeswitch-users at lists.freeswitch.org> 
Sent: Tue May 18 11:56:28 2010
Subject: Re: [Freeswitch-users] Contact Field 

HI Ken,

 

Add the following line

 

<param name="extension" value="10 Digit Number"/>

 

Lloyd



On Tue, May 18, 2010 at 11:40 AM, Ken Fulmer
<kenfulmer at icstechnologysolutions.com> wrote:

We are connecting a FS gateway to PaeTec. They need the contact field to be
the same as the calling number. Using the trunk registration 10 digit number
doesn't work for them.

 

I added the following parameter to remove "gw + external" in the contact
field to the left of the @ sign:

 

    <param name="extension-in-contact" value="true"/>

 

Is there something else I should add, so the contact field shows the 10
digit calling number? 

 

Thanks,

 

Ken Fulmer

 


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

 

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From anthony.minessale at gmail.com  Tue May 18 11:07:47 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 18 May 2010 13:07:47 -0500
Subject: [Freeswitch-users] Contact Field
In-Reply-To: <00cb01caf6b2$8ae3b1b0$a0ab1510$@com>
References: <0b56b97ea4e1c52f05eea74c215aeaf1@mail.gmail.com>
	<00cb01caf6b2$8ae3b1b0$a0ab1510$@com>
Message-ID: <AANLkTimHClO_jUm2qHUZTFPxI0GEoItviqI6XrONdWEa@mail.gmail.com>

the answer was supplied 2 or 3 emails ago

On Tue, May 18, 2010 at 12:50 PM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  Yes, I?m asking about the contact field in the INVITE header. PaeTec
> won?t accept our calls without this change.
>
>
>
> Thanks,
>
>
>
> Ken
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Kristian
> Kielhofner
> *Sent:* Tuesday, May 18, 2010 11:30 AM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Contact Field
>
>
>
> I assume he is talking about the contact field in the INVITE, not the
> REGISTER...
>
> I don't know how to accomplish this off the top of my head but I'd be
> interested too (quite a few carriers request this).
>
>
> --
> Kristian Kielhofner
> http://blog.krisk.org
>
>
>  ------------------------------
>
> *From*: freeswitch-users-bounces at lists.freeswitch.org <
> freeswitch-users-bounces at lists.freeswitch.org>
> *To*: freeswitch-users at lists.freeswitch.org <
> freeswitch-users at lists.freeswitch.org>
> *Sent*: Tue May 18 11:56:28 2010
> *Subject*: Re: [Freeswitch-users] Contact Field
>
> HI Ken,
>
>
>
> Add the following line
>
>
>
> <param name="extension" value="10 Digit Number"/>
>
>
>
> Lloyd
>
>  On Tue, May 18, 2010 at 11:40 AM, Ken Fulmer <
> kenfulmer at icstechnologysolutions.com> wrote:
>
> We are connecting a FS gateway to PaeTec. They need the contact field to be
> the same as the calling number. Using the trunk registration 10 digit number
> doesn?t work for them.
>
>
>
> I added the following parameter to remove ?gw + external? in the contact
> field to the left of the @ sign:
>
>
>
>     <param name="extension-in-contact" value="true"/>
>
>
>
> Is there something else I should add, so the contact field shows the 10
> digit calling number?
>
>
>
> Thanks,
>
>
>
> Ken Fulmer
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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From jcasale at activenetwerx.com  Tue May 18 11:47:17 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Tue, 18 May 2010 18:47:17 +0000
Subject: [Freeswitch-users] centos 5.4 timer issue 1000hz,
 choppy voice at times
In-Reply-To: <AANLkTikhNMygm-3Se-6_xYJ5XgZELe-unaBMzeRMlvQa@mail.gmail.com>
References: <AANLkTikhNMygm-3Se-6_xYJ5XgZELe-unaBMzeRMlvQa@mail.gmail.com>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E9A0E837@prato.activenetwerx.local>

>[root at pum2 ~]# ./timerchk?
>kernel timer interrupt frequency is approx. 1002 Hz

So Anthony,
What is the expected safe deviation fs would operate in where any
audio issues I could assume would not be related to a shady clock.

On a variety of platforms I see 2-3 Hz variation but only on some
does fs complain. Or could it be that this deviation alone isn't
enough to quantify the quality of the clock?

Thanks for any input!
jlc


From msc at freeswitch.org  Tue May 18 12:08:48 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 18 May 2010 12:08:48 -0700
Subject: [Freeswitch-users] Hangup cause when timeout
In-Reply-To: <AANLkTingRSRk852foYYQxxglRmLqUX4O3ElYjTXZAYsB@mail.gmail.com>
References: <AANLkTingRSRk852foYYQxxglRmLqUX4O3ElYjTXZAYsB@mail.gmail.com>
Message-ID: <AANLkTim9Epy7yeoxL9LWD1buILt7vbF-QcyTeu5nypy7@mail.gmail.com>

Check this out:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_respond

try respond with a 503.

-MC

On Tue, May 18, 2010 at 12:44 AM, Dome Charoenyost <dome at tel.co.th> wrote:

> Dear all,
>            I want to change hangup cause when bridge time out. now FS
> return. 19      480     (NO_ANSWER).
> I want to change to 34 503 because sender want to do fail over and try
> next route. so what's best way to do that ?
>
> modify sofia is my idea :)
> BG
> Dome C.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
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From kenfulmer at icstechnologysolutions.com  Tue May 18 12:09:24 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Tue, 18 May 2010 14:09:24 -0500
Subject: [Freeswitch-users] Contact Field
In-Reply-To: <AANLkTimHClO_jUm2qHUZTFPxI0GEoItviqI6XrONdWEa@mail.gmail.com>
References: <0b56b97ea4e1c52f05eea74c215aeaf1@mail.gmail.com>	<00cb01caf6b2$8ae3b1b0$a0ab1510$@com>
	<AANLkTimHClO_jUm2qHUZTFPxI0GEoItviqI6XrONdWEa@mail.gmail.com>
Message-ID: <00e801caf6bd$a16800c0$e4380240$@com>

So, do I add the following line verbatim or am I supposed to insert a real
10 digit number for the value?

 

<param name="extension" value="10 Digit Number"/>

 

Thanks,

 

Ken

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Tuesday, May 18, 2010 1:08 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Contact Field

 

the answer was supplied 2 or 3 emails ago

On Tue, May 18, 2010 at 12:50 PM, Ken Fulmer
<kenfulmer at icstechnologysolutions.com> wrote:

Yes, I'm asking about the contact field in the INVITE header. PaeTec won't
accept our calls without this change.

 

Thanks,

 

Ken

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kristian
Kielhofner
Sent: Tuesday, May 18, 2010 11:30 AM


To: freeswitch-users at lists.freeswitch.org

Subject: Re: [Freeswitch-users] Contact Field

 

I assume he is talking about the contact field in the INVITE, not the
REGISTER...

I don't know how to accomplish this off the top of my head but I'd be
interested too (quite a few carriers request this).


-- 
Kristian Kielhofner 
http://blog.krisk.org

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
<freeswitch-users-bounces at lists.freeswitch.org> 
To: freeswitch-users at lists.freeswitch.org
<freeswitch-users at lists.freeswitch.org> 
Sent: Tue May 18 11:56:28 2010
Subject: Re: [Freeswitch-users] Contact Field 

HI Ken,

 

Add the following line

 

<param name="extension" value="10 Digit Number"/>

 

Lloyd

On Tue, May 18, 2010 at 11:40 AM, Ken Fulmer
<kenfulmer at icstechnologysolutions.com> wrote:

We are connecting a FS gateway to PaeTec. They need the contact field to be
the same as the calling number. Using the trunk registration 10 digit number
doesn't work for them.

 

I added the following parameter to remove "gw + external" in the contact
field to the left of the @ sign:

 

    <param name="extension-in-contact" value="true"/>

 

Is there something else I should add, so the contact field shows the 10
digit calling number? 

 

Thanks,

 

Ken Fulmer

 


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

 


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




-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
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From anthony.minessale at gmail.com  Tue May 18 12:27:10 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 18 May 2010 14:27:10 -0500
Subject: [Freeswitch-users] centos 5.4 timer issue 1000hz,
	choppy voice at 	times
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E9A0E837@prato.activenetwerx.local>
References: <AANLkTikhNMygm-3Se-6_xYJ5XgZELe-unaBMzeRMlvQa@mail.gmail.com>
	<CA5A491E9DEFBE4CB777DE97E21575E9A0E837@prato.activenetwerx.local>
Message-ID: <AANLkTimjakBxVH6b0_3l48M8-Y_gy-zjw6-YSJCCmzT2@mail.gmail.com>

2-3 is fine.
but the problem is when it's actually put to the test in userspace where you
have many threads competing for cpu, that is where you have to measure the
accuracy which is what that test during startup is doing.  using FS to sleep
1000 microseconds and timing how close it actually is to that.

voip can withstand jitter on most implementation as long as the running
average of the packets is on the mark, the timing of each individual packet
is less important, this worsens over latent connections.

try this from the cli

time_test 1000

you will see it come out + or - 100 easily (there is overhead for the code
executing as well as other factors  that can skew the results)

then try it with a larger sample:

time_test 1000 1000
time_test 1000 2000
etc

the more times you try it the more likely the average in the long run will
be accurate which is all we can hope for in our pseudo real time
environment.

most times if you try this with -nocal you will always get 2000 when you
intended for 1000
which means all your 1ms sleeps are double length from what was requested.

some newer kernels try to make it more accurate but by doing so they eat up
so much cpu it's not worth it.  For the most part semi-accurate 1ms timing
is more than enough.







On Tue, May 18, 2010 at 1:47 PM, Joseph L. Casale <jcasale at activenetwerx.com
> wrote:

> >[root at pum2 ~]# ./timerchk
> >kernel timer interrupt frequency is approx. 1002 Hz
>
> So Anthony,
> What is the expected safe deviation fs would operate in where any
> audio issues I could assume would not be related to a shady clock.
>
> On a variety of platforms I see 2-3 Hz variation but only on some
> does fs complain. Or could it be that this deviation alone isn't
> enough to quantify the quality of the clock?
>
> Thanks for any input!
> jlc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <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>
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From msc at freeswitch.org  Tue May 18 12:27:32 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 18 May 2010 12:27:32 -0700
Subject: [Freeswitch-users] Contact Field
In-Reply-To: <00e801caf6bd$a16800c0$e4380240$@com>
References: <0b56b97ea4e1c52f05eea74c215aeaf1@mail.gmail.com>
	<00cb01caf6b2$8ae3b1b0$a0ab1510$@com>
	<AANLkTimHClO_jUm2qHUZTFPxI0GEoItviqI6XrONdWEa@mail.gmail.com>
	<00e801caf6bd$a16800c0$e4380240$@com>
Message-ID: <AANLkTikZCTEsD-VpPP0tDI8GIUyEDmlNewQiOdXehjZM@mail.gmail.com>

I'm guessing a real 10 digit number. (Some want 1 + ten digit number).
-MC

On Tue, May 18, 2010 at 12:09 PM, Ken Fulmer <
kenfulmer at icstechnologysolutions.com> wrote:

>  So, do I add the following line verbatim or am I supposed to insert a
> real 10 digit number for the value?
>
>
>
> <param name="extension" value="10 Digit Number"/>
>
>
>
> Thanks,
>
>
>
> Ken
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Tuesday, May 18, 2010 1:08 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Contact Field
>
>
>
> the answer was supplied 2 or 3 emails ago
>
> On Tue, May 18, 2010 at 12:50 PM, Ken Fulmer <
> kenfulmer at icstechnologysolutions.com> wrote:
>
> Yes, I?m asking about the contact field in the INVITE header. PaeTec won?t
> accept our calls without this change.
>
>
>
> Thanks,
>
>
>
> Ken
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Kristian
> Kielhofner
> *Sent:* Tuesday, May 18, 2010 11:30 AM
>
>
> *To:* freeswitch-users at lists.freeswitch.org
>
> *Subject:* Re: [Freeswitch-users] Contact Field
>
>
>
> I assume he is talking about the contact field in the INVITE, not the
> REGISTER...
>
> I don't know how to accomplish this off the top of my head but I'd be
> interested too (quite a few carriers request this).
>
>
> --
> Kristian Kielhofner
> http://blog.krisk.org
>
>
>  ------------------------------
>
> *From*: freeswitch-users-bounces at lists.freeswitch.org <
> freeswitch-users-bounces at lists.freeswitch.org>
> *To*: freeswitch-users at lists.freeswitch.org <
> freeswitch-users at lists.freeswitch.org>
> *Sent*: Tue May 18 11:56:28 2010
> *Subject*: Re: [Freeswitch-users] Contact Field
>
> HI Ken,
>
>
>
> Add the following line
>
>
>
> <param name="extension" value="10 Digit Number"/>
>
>
>
> Lloyd
>
> On Tue, May 18, 2010 at 11:40 AM, Ken Fulmer <
> kenfulmer at icstechnologysolutions.com> wrote:
>
> We are connecting a FS gateway to PaeTec. They need the contact field to be
> the same as the calling number. Using the trunk registration 10 digit number
> doesn?t work for them.
>
>
>
> I added the following parameter to remove ?gw + external? in the contact
> field to the left of the @ sign:
>
>
>
>     <param name="extension-in-contact" value="true"/>
>
>
>
> Is there something else I should add, so the contact field shows the 10
> digit calling number?
>
>
>
> Thanks,
>
>
>
> Ken Fulmer
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From wasim at convergence.pk  Tue May 18 12:31:28 2010
From: wasim at convergence.pk (Wasim Baig)
Date: Wed, 19 May 2010 00:31:28 +0500
Subject: [Freeswitch-users] centos 5.4 timer issue 1000hz,
	choppy voice at 	times
In-Reply-To: <AANLkTinAK8jYwaSdr4hcmv5j15Wj7HxHe5HqCV4x6Ss8@mail.gmail.com>
References: <AANLkTikhNMygm-3Se-6_xYJ5XgZELe-unaBMzeRMlvQa@mail.gmail.com> 
	<549CFEF87AEDE841A38E9D15EAB4C04C557D44DDE1@cooper>
	<AANLkTinAK8jYwaSdr4hcmv5j15Wj7HxHe5HqCV4x6Ss8@mail.gmail.com>
Message-ID: <AANLkTinoCG_REiKH59cbKJlHeVwAZ8fq0pLYFwKDgtaX@mail.gmail.com>

Just so others reading this thread years down the line don't get the wrong
idea, Tony's bark is much worse that his bite, he's really a big softie
underneath, going out of his way, leaving no stone unturned to help folks
out and ensure that any problem is identified quickly and accurately, and a
fix provided pronto, proof being the yummy pudding FS is.

Further comments inline below ...

On Tue, May 18, 2010 at 20:35, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Please heed as this is our official policy.
>
> There is not an issue with CentOS 5.3.
> In fact CentOS 5.3 is the preferred platform (not 5.4 or 5.5 until we do
> proper testing and QA).
>
What you are describing is the result of poor hardware, (most likely the
> motherboard)
>

If its the motherboard, then it should affect 5.3 as well.  If not, then the
issue is 5.4, and should (hopefully) be more addressable, rather than
chucking out the hardware.
For others with this issue, I'd recommend you drop down to 5.3 and verify
the HW, if it goes away with 5.3, then lets see whats needed to fix it in
5.4/5.5.

fwiw, i'm gonna try it on brand spanking new hpdl180g6 quad core xeon, will
try both 5.3 and 5.4

FreeSWITCH is Free, your box is not, use the money you save not buying
> FreeSWITCH to buy a decent box.  Even a $500 box can work perfectly fine.
>  Its not my intent to cater to inferior hardware when my goal is to push
> telephony into the future not get caught up in legacy drama with older
> boxes.
>

tony, these are the same boxes we did FS on way back when, remember the MGCP
UA mode for * would keep crashing and you said screw it and started writing
up FS instead,  that was ages ago ... at least 5 years back ... so yeh, old
boxes, dell poweredge 1450 ... point taken

If you have altered the kernel to run at something besides 1000hz, change it
> back to the default of 1000hz.   If you get the error instantly it means the
> kernel is actually configured at the wrong freq.
> If the test runs for a long time and concludes it, it most likely means the
> box is not good enough to run FS.
>

sometimes it runs for long, sometimes its real fast ...


> If you are not careful you may have installed the xen build of the kernel
> (check with uname -a)
> This can also cause timing issues.
>

stock kernel, 2.6.18-164.el5

If you see that timing gap warning (especially on centos if you have it set
> to 1000hz) it means the motherboard cannot give the OS a good enough timing
> source to provide 1ms resolution which is mandatory and not very demanding.
>
> I have been on dozens of perfectly happy boxes running CentOS 5.3 and I
> have been on a small handful of boxes that do what you describe and have
> seen that, once they are where they belong, (in the trash) the new
> inexpensive replacement does just fine.
>

:) ... 10-4

-wasim


> On Tue, May 18, 2010 at 2:58 AM, Peter Olsson <
> peter.olsson at visionutveckling.se> wrote:
>
>>  I also have this behaviour (timer gap around 2000). Upgraded yesterday
>> to CentOS 5.5 (from 5.4), but the issue still remains. The system is just
>> used for lab purposes, so I could easily fix SSH access if someone wanted to
>> have a closer look.
>>
>>
>>
>> I should also add ? this is a 3-4 year old machine (workstation), but it
>> seems this problem is related to newer CentOS-kernels, so I guess it
>> shouldn?t be the issue here.
>>
>>
>>
>> /Peter
>>
>>
>>
>> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *F?r *Wasim Baig
>> *Skickat:* den 18 maj 2010 06:20
>> *Till:* FreeSWITCH Users
>> *?mne:* [Freeswitch-users] centos 5.4 timer issue 1000hz, choppy voice at
>> times
>>
>>
>>
>> I know its been discussed previously on list, but has there been any
>> progress to the timer issue on centos 5.4 32 bit
>>
>>
>>
>> Linux pum2 2.6.18-164.el5 #1 SMP Thu Sep 3 03:33:56 EDT 2009 i686 i686
>> i386 GNU/Linux
>>
>>
>>
>> [root at pum2 ~]# grep 1000 /boot/config-2.6.18-164.el5
>>
>> CONFIG_HZ_1000=y
>>
>> CONFIG_HZ=1000
>>
>>
>>
>> i used a basic timer testing utility to confirm what we're actually
>> getting
>>
>>
>>
>> [root at pum2 ~]# ./timerchk
>>
>> kernel timer interrupt frequency is approx. 1002 Hz
>>
>>
>>
>> however, fs complains ...
>>
>>
>>
>> 2010-05-18 06:17:24.711360 [CONSOLE] switch_time.c:1032 Calibrating timer,
>> please wait...
>>
>> 2010-05-18 06:17:24.861063 [CONSOLE] switch_time.c:223 Test: 1000 Average:
>> 2992 Step: 50
>>
>> 2010-05-18 06:17:24.861097 [WARNING] switch_time.c:227 Abnormally large
>> timer gap 1992 detected!
>>
>> Do you have your kernel timer frequency set to lower than 1,000Hz? You may
>> experience audio problems.
>>
>>
>>
>> any recommendations, other than drop to 5.3? which is not doable as the
>> box is a 5k km away ...
>>
>>
>>
>> --
>> wasim h. baig | principal consultant | convergence pk | +92 30 0850 8070 |
>> peace be upon you ...
>>
>> !DSPAM:4bf2176432938985444358!
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
wasim h. baig | principal consultant | convergence pk | +92 30 0850 8070 |
peace be upon you ...
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From dftoro at yahoo.com  Tue May 18 12:32:44 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Tue, 18 May 2010 12:32:44 -0700 (PDT)
Subject: [Freeswitch-users] TLS on Windows
Message-ID: <215950.79428.qm@web33501.mail.mud.yahoo.com>

Greetings,

I have been interested about TLS on Windows and FreeSWITCH, after web search and mailing list, I don't know if is possible using FreeSWITCH on Windows to have security communication with TLS. 
 
I made and installed OpenSSL and I have tried build FreeSWITCH (libsofia_sip_ua_static) using OpenSSL, I created the certificates and put it on conf/ssl directory, but the internal profile faults when it's created, the sofia log says some like protocol is not supported.

I have lastest git branch 1.0.head, windows 7 32 bits.

I appreciate any suggestion

Thank you.

Diego Toro
http://lacarretade.blogspot.com/


      


From kenfulmer at icstechnologysolutions.com  Tue May 18 13:05:49 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Tue, 18 May 2010 15:05:49 -0500
Subject: [Freeswitch-users] Contact Field
In-Reply-To: <AANLkTikZCTEsD-VpPP0tDI8GIUyEDmlNewQiOdXehjZM@mail.gmail.com>
References: <0b56b97ea4e1c52f05eea74c215aeaf1@mail.gmail.com>	<00cb01caf6b2$8ae3b1b0$a0ab1510$@com>	<AANLkTimHClO_jUm2qHUZTFPxI0GEoItviqI6XrONdWEa@mail.gmail.com>	<00e801caf6bd$a16800c0$e4380240$@com>
	<AANLkTikZCTEsD-VpPP0tDI8GIUyEDmlNewQiOdXehjZM@mail.gmail.com>
Message-ID: <010d01caf6c5$8337e8b0$89a7ba10$@com>

The problem is, PaeTec wants to see the actual calling number in the contact
field, not the same number each time. 

 

Ken

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Tuesday, May 18, 2010 2:28 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Contact Field

 

I'm guessing a real 10 digit number. (Some want 1 + ten digit number).
-MC

On Tue, May 18, 2010 at 12:09 PM, Ken Fulmer
<kenfulmer at icstechnologysolutions.com> wrote:

So, do I add the following line verbatim or am I supposed to insert a real
10 digit number for the value?

 

<param name="extension" value="10 Digit Number"/>

 

Thanks,

 

Ken

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale
Sent: Tuesday, May 18, 2010 1:08 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Contact Field

 

the answer was supplied 2 or 3 emails ago

On Tue, May 18, 2010 at 12:50 PM, Ken Fulmer
<kenfulmer at icstechnologysolutions.com> wrote:

Yes, I'm asking about the contact field in the INVITE header. PaeTec won't
accept our calls without this change.

 

Thanks,

 

Ken

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kristian
Kielhofner
Sent: Tuesday, May 18, 2010 11:30 AM


To: freeswitch-users at lists.freeswitch.org

Subject: Re: [Freeswitch-users] Contact Field

 

I assume he is talking about the contact field in the INVITE, not the
REGISTER...

I don't know how to accomplish this off the top of my head but I'd be
interested too (quite a few carriers request this).


-- 
Kristian Kielhofner 
http://blog.krisk.org

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
<freeswitch-users-bounces at lists.freeswitch.org> 
To: freeswitch-users at lists.freeswitch.org
<freeswitch-users at lists.freeswitch.org> 
Sent: Tue May 18 11:56:28 2010
Subject: Re: [Freeswitch-users] Contact Field 

HI Ken,

 

Add the following line

 

<param name="extension" value="10 Digit Number"/>

 

Lloyd

On Tue, May 18, 2010 at 11:40 AM, Ken Fulmer
<kenfulmer at icstechnologysolutions.com> wrote:

We are connecting a FS gateway to PaeTec. They need the contact field to be
the same as the calling number. Using the trunk registration 10 digit number
doesn't work for them.

 

I added the following parameter to remove "gw + external" in the contact
field to the left of the @ sign:

 

    <param name="extension-in-contact" value="true"/>

 

Is there something else I should add, so the contact field shows the 10
digit calling number? 

 

Thanks,

 

Ken Fulmer

 


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

 


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




-- 
Anthony Minessale II

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

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

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


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

 

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From jan.berger at video24.no  Tue May 18 13:07:36 2010
From: jan.berger at video24.no (Jan Berger)
Date: Tue, 18 May 2010 22:07:36 +0200
Subject: [Freeswitch-users] SCTP for Windows
In-Reply-To: <20100518164235.3F93FD6247B00@bmail02.one.com>
References: <527B81D4113C455A964AAE20C28C3D49@dell9400>
	<B80AF699-A323-4031-8900-3AD3081912CB@jerris.com>
	<20100518164235.3F93FD6247B00@bmail02.one.com>
Message-ID: <9FFF1D7C542C46A2B98D5B8E787E7E4A@dell9400>

You need to download this from sourceforge so you get the english install,
newest version and doc.

 

The source I looked at say this is a Cisco sctp stack ???

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jan
Berger
Sent: 18. mai 2010 18:43
To: freeswitch-users at lists.freeswitch.org;
freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] SCTP for Windows

 

Thanks mate,

 

Japanese ? Might actually work then :)

 

There is a German library as well, but it says GPL. 

 

Jan 

On May 18, 2010 16:42 "Michael Jerris"  <mailto:mike at jerris.com>
<mike at jerris.com> wrote:

http://www.co-conv.jp/en/product/sctpDrv/20081224/ 

 

On May 18, 2010, at 8:09 AM, Jan Berger wrote:





Hi list,

 

Does anyone know about a proper SCTP library for Windows that is reliable
and with a compatible licens?

 

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From ranjtech at gmail.com  Tue May 18 13:14:18 2010
From: ranjtech at gmail.com (RR)
Date: Tue, 18 May 2010 16:14:18 -0400
Subject: [Freeswitch-users] FS on Sparc based Architecture
Message-ID: <AANLkTiklyJ-DwEyRxcEwd2mpvDqrBucZANCR3U2bGuFh@mail.gmail.com>

Hello List,

I did a bit of hunting here and there but couldn't find anything that
mentions any one having any experience/success/failure with
compiling/building FS under Solaris (other than
http://wiki.freeswitch.org/wiki/Installation_Guide#Solaris) running over a
Sparc based architecture. For all I know, it may have nothing to do with it
being Solaris on Sparc or Intel but does anyone know and can verify for sure
that using the instructions on the Wiki, FS can be built for Solaris on
Sparc and will/does stable and seamlessly as it does on Linux etc?

Sorry if this is a silly question but just needed to ask for my own sanity
:)
Thanks so much
RR
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From david.ponzone at gmail.com  Tue May 18 13:18:16 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Tue, 18 May 2010 22:18:16 +0200
Subject: [Freeswitch-users] Contact Field
In-Reply-To: <010d01caf6c5$8337e8b0$89a7ba10$@com>
References: <0b56b97ea4e1c52f05eea74c215aeaf1@mail.gmail.com>	<00cb01caf6b2$8ae3b1b0$a0ab1510$@com>	<AANLkTimHClO_jUm2qHUZTFPxI0GEoItviqI6XrONdWEa@mail.gmail.com>	<00e801caf6bd$a16800c0$e4380240$@com>
	<AANLkTikZCTEsD-VpPP0tDI8GIUyEDmlNewQiOdXehjZM@mail.gmail.com>
	<010d01caf6c5$8337e8b0$89a7ba10$@com>
Message-ID: <2D05B2FC-A5A1-41D0-AF65-5A4CE2869EE3@gmail.com>

Ken,

perhaps you can try to export sip_contact_user on leg B with the right  
value ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 18/05/2010 ? 22:05, Ken Fulmer a ?crit :

> The problem is, PaeTec wants to see the actual calling number in the  
> contact field, not the same number each time.
>
> Ken
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Michael Collins
> Sent: Tuesday, May 18, 2010 2:28 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Contact Field
>
> I'm guessing a real 10 digit number. (Some want 1 + ten digit number).
> -MC
>
> On Tue, May 18, 2010 at 12:09 PM, Ken Fulmer <kenfulmer at icstechnologysolutions.com 
> > wrote:
> So, do I add the following line verbatim or am I supposed to insert  
> a real 10 digit number for the value?
>
> <param name="extension" value="10 Digit Number"/>
>
> Thanks,
>
> Ken
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Anthony Minessale
> Sent: Tuesday, May 18, 2010 1:08 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Contact Field
>
> the answer was supplied 2 or 3 emails ago
>
> On Tue, May 18, 2010 at 12:50 PM, Ken Fulmer <kenfulmer at icstechnologysolutions.com 
> > wrote:
> Yes, I?m asking about the contact field in the INVITE header.  
> PaeTec won?t accept our calls without this change.
>
> Thanks,
>
> Ken
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Kristian Kielhofner
> Sent: Tuesday, May 18, 2010 11:30 AM
>
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Contact Field
>
> I assume he is talking about the contact field in the INVITE, not  
> the REGISTER...
>
> I don't know how to accomplish this off the top of my head but I'd  
> be interested too (quite a few carriers request this).
>
>
> -- 
> Kristian Kielhofner
> http://blog.krisk.org
>
> From: freeswitch-users-bounces at lists.freeswitch.org <freeswitch-users-bounces at lists.freeswitch.org 
> >
> To: freeswitch-users at lists.freeswitch.org <freeswitch-users at lists.freeswitch.org 
> >
> Sent: Tue May 18 11:56:28 2010
> Subject: Re: [Freeswitch-users] Contact Field
>
> HI Ken,
>
> Add the following line
>
> <param name="extension" value="10 Digit Number"/>
>
> Lloyd
>
> On Tue, May 18, 2010 at 11:40 AM, Ken Fulmer <kenfulmer at icstechnologysolutions.com 
> > wrote:
> We are connecting a FS gateway to PaeTec. They need the contact  
> field to be the same as the calling number. Using the trunk  
> registration 10 digit number doesn?t work for them.
>
> I added the following parameter to remove ?gw + external? in the  
> contact field to the left of the @ sign:
>
>     <param name="extension-in-contact" value="true"/>
>
> Is there something else I should add, so the contact field shows the  
> 10 digit calling number?
>
> Thanks,
>
> Ken Fulmer
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Tue May 18 13:28:09 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 18 May 2010 15:28:09 -0500
Subject: [Freeswitch-users] Contact Field
In-Reply-To: <2D05B2FC-A5A1-41D0-AF65-5A4CE2869EE3@gmail.com>
References: <0b56b97ea4e1c52f05eea74c215aeaf1@mail.gmail.com>	<00cb01caf6b2$8ae3b1b0$a0ab1510$@com>	<AANLkTimHClO_jUm2qHUZTFPxI0GEoItviqI6XrONdWEa@mail.gmail.com>	<00e801caf6bd$a16800c0$e4380240$@com>
	<AANLkTikZCTEsD-VpPP0tDI8GIUyEDmlNewQiOdXehjZM@mail.gmail.com>
	<010d01caf6c5$8337e8b0$89a7ba10$@com>
	<2D05B2FC-A5A1-41D0-AF65-5A4CE2869EE3@gmail.com>
Message-ID: <68AE9BBA-73C5-41FF-BF6C-F5A21E7456BC@freeswitch.org>

Really we are STILL TALKING about this? 

/b

On May 18, 2010, at 3:18 PM, David Ponzone wrote:

> Ken,
> 
> perhaps you can try to export sip_contact_user on leg B with the right value ?
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com

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From leo.zibi at gmail.com  Tue May 18 13:38:41 2010
From: leo.zibi at gmail.com (leo.zibi at gmail.com)
Date: Tue, 18 May 2010 22:38:41 +0200
Subject: [Freeswitch-users] TLS on Windows
In-Reply-To: <215950.79428.qm@web33501.mail.mud.yahoo.com>
References: <215950.79428.qm@web33501.mail.mud.yahoo.com>
Message-ID: <4BF2FAD1.1030009@gmail.com>

Hi,

http://wiki.freeswitch.org/wiki/Generating_TLS_certificates_(win32)

-- 
Regards
P.

Diego Toro wrote:
> Greetings,
>
> I have been interested about TLS on Windows and FreeSWITCH, after web search and mailing list, I don't know if is possible using FreeSWITCH on Windows to have security communication with TLS. 
>  
> I made and installed OpenSSL and I have tried build FreeSWITCH (libsofia_sip_ua_static) using OpenSSL, I created the certificates and put it on conf/ssl directory, but the internal profile faults when it's created, the sofia log says some like protocol is not supported.
>
> I have lastest git branch 1.0.head, windows 7 32 bits.
>
> I appreciate any suggestion
>
> Thank you.
>
> Diego Toro
> http://lacarretade.blogspot.com/
>
>
>       
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   



From kenfulmer at icstechnologysolutions.com  Tue May 18 13:52:14 2010
From: kenfulmer at icstechnologysolutions.com (Ken Fulmer)
Date: Tue, 18 May 2010 15:52:14 -0500
Subject: [Freeswitch-users] Contact Field
In-Reply-To: <68AE9BBA-73C5-41FF-BF6C-F5A21E7456BC@freeswitch.org>
References: <0b56b97ea4e1c52f05eea74c215aeaf1@mail.gmail.com>	<00cb01caf6b2$8ae3b1b0$a0ab1510$@com>	<AANLkTimHClO_jUm2qHUZTFPxI0GEoItviqI6XrONdWEa@mail.gmail.com>	<00e801caf6bd$a16800c0$e4380240$@com>	<AANLkTikZCTEsD-VpPP0tDI8GIUyEDmlNewQiOdXehjZM@mail.gmail.com>	<010d01caf6c5$8337e8b0$89a7ba10$@com>	<2D05B2FC-A5A1-41D0-AF65-5A4CE2869EE3@gmail.com>
	<68AE9BBA-73C5-41FF-BF6C-F5A21E7456BC@freeswitch.org>
Message-ID: <013c01caf6cb$ff1264f0$fd372ed0$@com>

Is it a taboo subject?

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
West
Sent: Tuesday, May 18, 2010 3:28 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Contact Field

 

Really we are STILL TALKING about this? 

 

/b

 

On May 18, 2010, at 3:18 PM, David Ponzone wrote:





Ken,

 

perhaps you can try to export sip_contact_user on leg B with the right value
?

 

David Ponzone  Direction Technique

email: david.ponzone at ipeva.fr

tel:      01 74 03 18 97

gsm:   06 66 98 76 34

 

Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>   -   www.ipeva-studio.com
<BLOCKED::http://www.ipeva-studio.com/> 

 

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From brian at freeswitch.org  Tue May 18 13:56:33 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 18 May 2010 15:56:33 -0500
Subject: [Freeswitch-users] Contact Field
In-Reply-To: <013c01caf6cb$ff1264f0$fd372ed0$@com>
References: <0b56b97ea4e1c52f05eea74c215aeaf1@mail.gmail.com>	<00cb01caf6b2$8ae3b1b0$a0ab1510$@com>	<AANLkTimHClO_jUm2qHUZTFPxI0GEoItviqI6XrONdWEa@mail.gmail.com>	<00e801caf6bd$a16800c0$e4380240$@com>	<AANLkTikZCTEsD-VpPP0tDI8GIUyEDmlNewQiOdXehjZM@mail.gmail.com>	<010d01caf6c5$8337e8b0$89a7ba10$@com>	<2D05B2FC-A5A1-41D0-AF65-5A4CE2869EE3@gmail.com>
	<68AE9BBA-73C5-41FF-BF6C-F5A21E7456BC@freeswitch.org>
	<013c01caf6cb$ff1264f0$fd372ed0$@com>
Message-ID: <FEBED172-FA3F-4A08-A202-6068D4050DB1@freeswitch.org>

I'm just sure the dead horse has been beat to ground meat.

/b

On May 18, 2010, at 3:52 PM, Ken Fulmer wrote:

> Is it a taboo subject?
>  
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
> Sent: Tuesday, May 18, 2010 3:28 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Contact Field
>  
> Really we are STILL TALKING about this? 
>  
> /b

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From grsingh750 at gmail.com  Tue May 18 14:26:12 2010
From: grsingh750 at gmail.com (guru singh)
Date: Wed, 19 May 2010 02:56:12 +0530
Subject: [Freeswitch-users] Hardware Suggestions. (David Ponzone)
Message-ID: <AANLkTikeoUKdg6uYKgTi9CDFpEO_lP9XNUlywnrjpx34@mail.gmail.com>

(My bad, I somehow missed editing the subject line.)
Thanks for the Zeroshell link, I'm having a look at it. Vyatta may've
been overkill for my setup.

gs

> From:?David Ponzone <david.ponzone at gmail.com>
> To:?freeswitch-users at lists.freeswitch.org
> Date:?Tue, 18 May 2010 19:26:23 +0200
> Subject:?Re: [Freeswitch-users] Hardware Suggestions.
> (sorry for this repost, but the previous one was by mistake sent with the wrong subject, which was bad for the readability of this thread).
> Guru,
> you may find a refurbished Cisco "POE" switch :)
> For the router part, you may also check Zeroshell (http://www.zeroshell.net/eng/).
> It's more CPE-oriented and more multi-purpose than Vyatta.
> David Ponzone ?Direction Technique
> email: david.ponzone at ipeva.fr
> tel: ? ? ?01 74 03 18 97
> gsm: ? 06 66 98 76 34
> Service Client?IPeva
> tel: ? ? ?0811 46 26 26
> www.ipeva.fr? -? ?www.ipeva-studio.com


From freeswitch at gilligan.id.au  Tue May 18 15:04:18 2010
From: freeswitch at gilligan.id.au (Chris)
Date: Wed, 19 May 2010 08:04:18 +1000
Subject: [Freeswitch-users] mod_managed or a script,
	sending DTMF and 	receiving DTMF
In-Reply-To: <00054046A7FA4FE59A5C5AB41C43382C@MOBILEE1705>
References: <AANLkTikSTwOw1yiBGO9VrVDnsKtH7ySz8V2LxL1XUdru@mail.gmail.com> 
	<AANLkTimDZx2T15GHnWck36bESESuK8n4u_wzzujyxvfE@mail.gmail.com> 
	<AANLkTilXbUSK5DbDb5RWboqEJLB6xAuPVwnDA21KsuZK@mail.gmail.com> 
	<B429D30727D34D35B0FA648556D337F3@MOBILEE1705>
	<AANLkTimGSAW5M9dmu2r70M2AM6jRMcSWZxjA__K7h8XW@mail.gmail.com> 
	<7350123A2AAC421587893A6BD6657C74@MOBILEE1705>
	<AANLkTilXDGyz4MUWg3Mji1dKkeWCLKsRcd3JNPvZd97t@mail.gmail.com> 
	<3D1F00DE01404A7E9D9C1942D401B982@MOBILEE1705>
	<AANLkTilIvDqbZ0P1PKDo7z_79H8bIUP6Y_7BkMLubKws@mail.gmail.com> 
	<00054046A7FA4FE59A5C5AB41C43382C@MOBILEE1705>
Message-ID: <AANLkTikIT4mKWkAkd0h3eJQipp7Q0eCQLYxyHZi7O9KN@mail.gmail.com>

Hi,
ok i have fixed most of my DTMF issues now but i have one provider that is
still stuffed. I receive 2833 and info perfectly from them but 2833 sent to
them mostly stuffs up and INFO stuffs up if it is the same digits sent 3
times or some times randomly.

My thoughts are they run both new and old asterisk servers and the issue
could be between their own servers.  Either way i need to prove to them it
is not my server by sending via in-band DTMF as i use u-law as my server is
in their Datacenter.

The issue is i am not sure how to turn in-band on.  I mean the sip-profile
has a comment there to only set 2833 or INFO there and the other to do via
the dial plan but all my tones etc are going via the console and mod_managed
so i am a little confused on the best way to do this.

Regards

Chris

On Sat, May 15, 2010 at 12:18 PM, Madovsky <infos at madovsky.org> wrote:

>  you can set it into your sip_profiles
>
> ----- Original Message -----
> *From:* Chris <freeswitch at gilligan.id.au>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Friday, May 14, 2010 10:11 PM
> *Subject:* Re: [Freeswitch-users] mod_managed or a script,sending DTMF and
> receiving DTMF
>
> would you happen to know the default one in freeswitch? My system is for
> landlines anyway and that seems to work.  the voip phone was just for free
> testing so i did not have to pay for calls.
>
> On Sat, May 15, 2010 at 11:59 AM, Madovsky <infos at madovsky.org> wrote:
>
>>  read carefully RFC2833, RFC2976 and inband DTMF.
>> there are 3 major different ways to use DTMF. read the doc of your phone
>> to know which DTMF it uses
>>
>> F
>>
>>  ----- Original Message -----
>> *From:* Chris <freeswitch at gilligan.id.au>
>> *To:* freeswitch-users at lists.freeswitch.org
>>   *Sent:* Friday, May 14, 2010 9:45 PM
>> *Subject:* Re: [Freeswitch-users] mod_managed or a script,sending DTMF
>> and receiving DTMF
>>
>> ok worked out the problem. Thanks for the help.  my stupid softphone does
>> not support dtmf receiving ti seems.  used normal phone to call and it all
>> worked.  grrrr
>>
>> On Sat, May 15, 2010 at 11:39 AM, Madovsky <infos at madovsky.org> wrote:
>>
>>>  I use it with fs_cli
>>>
>>>   ----- Original Message -----
>>> *From:* Chris <freeswitch at gilligan.id.au>
>>> *To:* freeswitch-users at lists.freeswitch.org
>>> *Sent:* Friday, May 14, 2010 9:33 PM
>>> *Subject:* Re: [Freeswitch-users] mod_managed or a script,sending DTMF
>>> and receiving DTMF
>>>
>>> do you happen to be using ti from inside mod_managed or via a different
>>> means?  Any chance of a small sample?  maybe i am doing something stupid.
>>>
>>> On Sat, May 15, 2010 at 11:22 AM, Madovsky <infos at madovsky.org> wrote:
>>>
>>>>  it works, I use it everyday
>>>>
>>>>   ----- Original Message -----
>>>> *From:* Chris <freeswitch at gilligan.id.au>
>>>> *To:* freeswitch-users at lists.freeswitch.org
>>>> *Sent:* Friday, May 14, 2010 8:52 PM
>>>> *Subject:* Re: [Freeswitch-users] mod_managed or a script,sending DTMF
>>>> and receiving DTMF
>>>>
>>>> sadly uuid_send_dtmf did not seem to work
>>>>
>>>>
>>>> On Fri, May 14, 2010 at 11:10 AM, Chris <freeswitch at gilligan.id.au>wrote:
>>>>
>>>>> Thanks so that is the only way?  I will have to do some tests and see
>>>>> if i can capture dtmf at the same time as i send when i do it that way.  Any
>>>>> other options?
>>>>>
>>>>>
>>>>> On Thu, May 13, 2010 at 10:50 PM, Phillip Jones <pjintheusa at gmail.com>wrote:
>>>>>
>>>>>> Try using http://wiki.freeswitch.org/wiki/Mod_commands#uuid_send_dtmf
>>>>>>
>>>>>>   On Thu, May 13, 2010 at 5:32 AM, Chris <freeswitch at gilligan.id.au>wrote:
>>>>>>
>>>>>>>  HI,
>>>>>>> I am currently working on a project in mod_managed and i am trying to
>>>>>>> discover the best way to meet my requirements.  i am hoping someone will
>>>>>>> have some ideas.  This could be implemented in one of the other scripting
>>>>>>> language if needed.
>>>>>>>
>>>>>>> What i am trying to do is reprogram a remote device via the phone.
>>>>>>>  The device takes commands in the form of DTMF tones and responds in
>>>>>>> different DTMF tones depending on success or failure. An example of the flow
>>>>>>> would be
>>>>>>>
>>>>>>> freeswitch -> 342523#
>>>>>>> device -> 1
>>>>>>> freeswitch -> 356789#
>>>>>>> device ->2
>>>>>>>
>>>>>>> device always responds with one digit and freeswitch sends many.
>>>>>>>
>>>>>>> I was hoping to do this with a database and mod_managed but i can't
>>>>>>> workout how to send the DTMF in mod_managed unless i user audio files for it
>>>>>>> which seems to be the wrong way to go to me. While i would prefer a
>>>>>>> mod_managed solution i will take anything i can find.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> Chris
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>  ------------------------------
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>  ------------------------------
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>  ------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dftoro at yahoo.com  Tue May 18 15:18:47 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Tue, 18 May 2010 15:18:47 -0700 (PDT)
Subject: [Freeswitch-users] TLS on Windows
In-Reply-To: <4BF2FAD1.1030009@gmail.com>
Message-ID: <746307.1047.qm@web33502.mail.mud.yahoo.com>

I have the certificates, this isn't my problem.  Sofia says some like protocol is not supported.

Thank you

Diego Toro
http://lacarretade.blogspot.com/


--- On Tue, 5/18/10, leo.zibi at gmail.com <leo.zibi at gmail.com> wrote:

> From: leo.zibi at gmail.com <leo.zibi at gmail.com>
> Subject: Re: [Freeswitch-users] TLS on Windows
> To: freeswitch-users at lists.freeswitch.org
> Date: Tuesday, May 18, 2010, 3:38 PM
> Hi,
> 
> http://wiki.freeswitch.org/wiki/Generating_TLS_certificates_(win32)
> 
> -- 
> Regards
> P.
> 
> Diego Toro wrote:
> > Greetings,
> >
> > I have been interested about TLS on Windows and
> FreeSWITCH, after web search and mailing list, I don't know
> if is possible using FreeSWITCH on Windows to have security
> communication with TLS. 
> >? 
> > I made and installed OpenSSL and I have tried build
> FreeSWITCH (libsofia_sip_ua_static) using OpenSSL, I created
> the certificates and put it on conf/ssl directory, but the
> internal profile faults when it's created, the sofia log
> says some like protocol is not supported.
> >
> > I have lastest git branch 1.0.head, windows 7 32
> bits.
> >
> > I appreciate any suggestion
> >
> > Thank you.
> >
> > Diego Toro
> > http://lacarretade.blogspot.com/
> >
> >
> >? ? ???
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >???
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


      


From msc at freeswitch.org  Tue May 18 15:32:24 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 18 May 2010 15:32:24 -0700
Subject: [Freeswitch-users] Contact Field
In-Reply-To: <FEBED172-FA3F-4A08-A202-6068D4050DB1@freeswitch.org>
References: <0b56b97ea4e1c52f05eea74c215aeaf1@mail.gmail.com>
	<00cb01caf6b2$8ae3b1b0$a0ab1510$@com>
	<AANLkTimHClO_jUm2qHUZTFPxI0GEoItviqI6XrONdWEa@mail.gmail.com>
	<00e801caf6bd$a16800c0$e4380240$@com>
	<AANLkTikZCTEsD-VpPP0tDI8GIUyEDmlNewQiOdXehjZM@mail.gmail.com>
	<010d01caf6c5$8337e8b0$89a7ba10$@com>
	<2D05B2FC-A5A1-41D0-AF65-5A4CE2869EE3@gmail.com>
	<68AE9BBA-73C5-41FF-BF6C-F5A21E7456BC@freeswitch.org>
	<013c01caf6cb$ff1264f0$fd372ed0$@com>
	<FEBED172-FA3F-4A08-A202-6068D4050DB1@freeswitch.org>
Message-ID: <AANLkTim_VdJ55CutYBB8QN3hhF-m7IABnun4M-iZdNGk@mail.gmail.com>

On Tue, May 18, 2010 at 1:56 PM, Brian West <brian at freeswitch.org> wrote:

> I'm just sure the dead horse has been beat to ground meat.
>
> I missed the actual solution amidst the carnage! I searched our wiki for
paetec and didn't find anything. If someone can confirm the solution I will
confer w/ Ken and make sure it gets added to the interop section of our
wiki.
-MC
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From msc at freeswitch.org  Tue May 18 15:35:18 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 18 May 2010 15:35:18 -0700
Subject: [Freeswitch-users] mod_managed or a script,
	sending DTMF and 	receiving DTMF
In-Reply-To: <AANLkTikIT4mKWkAkd0h3eJQipp7Q0eCQLYxyHZi7O9KN@mail.gmail.com>
References: <AANLkTikSTwOw1yiBGO9VrVDnsKtH7ySz8V2LxL1XUdru@mail.gmail.com>
	<AANLkTilXbUSK5DbDb5RWboqEJLB6xAuPVwnDA21KsuZK@mail.gmail.com>
	<B429D30727D34D35B0FA648556D337F3@MOBILEE1705>
	<AANLkTimGSAW5M9dmu2r70M2AM6jRMcSWZxjA__K7h8XW@mail.gmail.com>
	<7350123A2AAC421587893A6BD6657C74@MOBILEE1705>
	<AANLkTilXDGyz4MUWg3Mji1dKkeWCLKsRcd3JNPvZd97t@mail.gmail.com>
	<3D1F00DE01404A7E9D9C1942D401B982@MOBILEE1705>
	<AANLkTilIvDqbZ0P1PKDo7z_79H8bIUP6Y_7BkMLubKws@mail.gmail.com>
	<00054046A7FA4FE59A5C5AB41C43382C@MOBILEE1705>
	<AANLkTikIT4mKWkAkd0h3eJQipp7Q0eCQLYxyHZi7O9KN@mail.gmail.com>
Message-ID: <AANLkTikBrgtH1iMMoCsoThTYTSl5Vv0N4WintA4QYkK9@mail.gmail.com>

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

On Tue, May 18, 2010 at 3:04 PM, Chris <freeswitch at gilligan.id.au> wrote:

> Hi,
> ok i have fixed most of my DTMF issues now but i have one provider that is
> still stuffed. I receive 2833 and info perfectly from them but 2833 sent to
> them mostly stuffs up and INFO stuffs up if it is the same digits sent 3
> times or some times randomly.
>
> My thoughts are they run both new and old asterisk servers and the issue
> could be between their own servers.  Either way i need to prove to them it
> is not my server by sending via in-band DTMF as i use u-law as my server is
> in their Datacenter.
>
> The issue is i am not sure how to turn in-band on.  I mean the sip-profile
> has a comment there to only set 2833 or INFO there and the other to do via
> the dial plan but all my tones etc are going via the console and mod_managed
> so i am a little confused on the best way to do this.
>
> Regards
>
> Chris
>
> On Sat, May 15, 2010 at 12:18 PM, Madovsky <infos at madovsky.org> wrote:
>
>>  you can set it into your sip_profiles
>>
>> ----- Original Message -----
>> *From:* Chris <freeswitch at gilligan.id.au>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Sent:* Friday, May 14, 2010 10:11 PM
>> *Subject:* Re: [Freeswitch-users] mod_managed or a script,sending DTMF
>> and receiving DTMF
>>
>> would you happen to know the default one in freeswitch? My system is for
>> landlines anyway and that seems to work.  the voip phone was just for free
>> testing so i did not have to pay for calls.
>>
>> On Sat, May 15, 2010 at 11:59 AM, Madovsky <infos at madovsky.org> wrote:
>>
>>>  read carefully RFC2833, RFC2976 and inband DTMF.
>>> there are 3 major different ways to use DTMF. read the doc of your phone
>>> to know which DTMF it uses
>>>
>>> F
>>>
>>>  ----- Original Message -----
>>> *From:* Chris <freeswitch at gilligan.id.au>
>>> *To:* freeswitch-users at lists.freeswitch.org
>>>   *Sent:* Friday, May 14, 2010 9:45 PM
>>> *Subject:* Re: [Freeswitch-users] mod_managed or a script,sending DTMF
>>> and receiving DTMF
>>>
>>> ok worked out the problem. Thanks for the help.  my stupid softphone does
>>> not support dtmf receiving ti seems.  used normal phone to call and it all
>>> worked.  grrrr
>>>
>>> On Sat, May 15, 2010 at 11:39 AM, Madovsky <infos at madovsky.org> wrote:
>>>
>>>>  I use it with fs_cli
>>>>
>>>>   ----- Original Message -----
>>>> *From:* Chris <freeswitch at gilligan.id.au>
>>>> *To:* freeswitch-users at lists.freeswitch.org
>>>> *Sent:* Friday, May 14, 2010 9:33 PM
>>>> *Subject:* Re: [Freeswitch-users] mod_managed or a script,sending DTMF
>>>> and receiving DTMF
>>>>
>>>> do you happen to be using ti from inside mod_managed or via a different
>>>> means?  Any chance of a small sample?  maybe i am doing something stupid.
>>>>
>>>> On Sat, May 15, 2010 at 11:22 AM, Madovsky <infos at madovsky.org> wrote:
>>>>
>>>>>  it works, I use it everyday
>>>>>
>>>>>   ----- Original Message -----
>>>>> *From:* Chris <freeswitch at gilligan.id.au>
>>>>> *To:* freeswitch-users at lists.freeswitch.org
>>>>> *Sent:* Friday, May 14, 2010 8:52 PM
>>>>> *Subject:* Re: [Freeswitch-users] mod_managed or a script,sending DTMF
>>>>> and receiving DTMF
>>>>>
>>>>> sadly uuid_send_dtmf did not seem to work
>>>>>
>>>>>
>>>>> On Fri, May 14, 2010 at 11:10 AM, Chris <freeswitch at gilligan.id.au>wrote:
>>>>>
>>>>>> Thanks so that is the only way?  I will have to do some tests and see
>>>>>> if i can capture dtmf at the same time as i send when i do it that way.  Any
>>>>>> other options?
>>>>>>
>>>>>>
>>>>>> On Thu, May 13, 2010 at 10:50 PM, Phillip Jones <pjintheusa at gmail.com
>>>>>> > wrote:
>>>>>>
>>>>>>> Try using
>>>>>>> http://wiki.freeswitch.org/wiki/Mod_commands#uuid_send_dtmf
>>>>>>>
>>>>>>>   On Thu, May 13, 2010 at 5:32 AM, Chris <freeswitch at gilligan.id.au>wrote:
>>>>>>>
>>>>>>>>  HI,
>>>>>>>> I am currently working on a project in mod_managed and i am trying
>>>>>>>> to discover the best way to meet my requirements.  i am hoping someone will
>>>>>>>> have some ideas.  This could be implemented in one of the other scripting
>>>>>>>> language if needed.
>>>>>>>>
>>>>>>>> What i am trying to do is reprogram a remote device via the phone.
>>>>>>>>  The device takes commands in the form of DTMF tones and responds in
>>>>>>>> different DTMF tones depending on success or failure. An example of the flow
>>>>>>>> would be
>>>>>>>>
>>>>>>>> freeswitch -> 342523#
>>>>>>>> device -> 1
>>>>>>>> freeswitch -> 356789#
>>>>>>>> device ->2
>>>>>>>>
>>>>>>>> device always responds with one digit and freeswitch sends many.
>>>>>>>>
>>>>>>>> I was hoping to do this with a database and mod_managed but i can't
>>>>>>>> workout how to send the DTMF in mod_managed unless i user audio files for it
>>>>>>>> which seems to be the wrong way to go to me. While i would prefer a
>>>>>>>> mod_managed solution i will take anything i can find.
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> Chris
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>  ------------------------------
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>  ------------------------------
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>  ------------------------------
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>  ------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From freeswitch at gilligan.id.au  Tue May 18 15:56:24 2010
From: freeswitch at gilligan.id.au (Chris)
Date: Wed, 19 May 2010 08:56:24 +1000
Subject: [Freeswitch-users] mod_managed or a script,
	sending DTMF and 	receiving DTMF
In-Reply-To: <AANLkTikBrgtH1iMMoCsoThTYTSl5Vv0N4WintA4QYkK9@mail.gmail.com>
References: <AANLkTikSTwOw1yiBGO9VrVDnsKtH7ySz8V2LxL1XUdru@mail.gmail.com> 
	<B429D30727D34D35B0FA648556D337F3@MOBILEE1705>
	<AANLkTimGSAW5M9dmu2r70M2AM6jRMcSWZxjA__K7h8XW@mail.gmail.com> 
	<7350123A2AAC421587893A6BD6657C74@MOBILEE1705>
	<AANLkTilXDGyz4MUWg3Mji1dKkeWCLKsRcd3JNPvZd97t@mail.gmail.com> 
	<3D1F00DE01404A7E9D9C1942D401B982@MOBILEE1705>
	<AANLkTilIvDqbZ0P1PKDo7z_79H8bIUP6Y_7BkMLubKws@mail.gmail.com> 
	<00054046A7FA4FE59A5C5AB41C43382C@MOBILEE1705>
	<AANLkTikIT4mKWkAkd0h3eJQipp7Q0eCQLYxyHZi7O9KN@mail.gmail.com> 
	<AANLkTikBrgtH1iMMoCsoThTYTSl5Vv0N4WintA4QYkK9@mail.gmail.com>
Message-ID: <AANLkTinmrIAFa43XVp0Kh1i7t4dcDCemtVA4wFnIc_kU@mail.gmail.com>

Thanks so much Michael that is exactly what i was looking for.  it is even
better than what i was expecting to find!

On Wed, May 19, 2010 at 8:35 AM, Michael Collins <msc at freeswitch.org> wrote:

> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf_generate
>
>
> On Tue, May 18, 2010 at 3:04 PM, Chris <freeswitch at gilligan.id.au> wrote:
>
>> Hi,
>> ok i have fixed most of my DTMF issues now but i have one provider that is
>> still stuffed. I receive 2833 and info perfectly from them but 2833 sent to
>> them mostly stuffs up and INFO stuffs up if it is the same digits sent 3
>> times or some times randomly.
>>
>> My thoughts are they run both new and old asterisk servers and the issue
>> could be between their own servers.  Either way i need to prove to them it
>> is not my server by sending via in-band DTMF as i use u-law as my server is
>> in their Datacenter.
>>
>> The issue is i am not sure how to turn in-band on.  I mean the sip-profile
>> has a comment there to only set 2833 or INFO there and the other to do via
>> the dial plan but all my tones etc are going via the console and mod_managed
>> so i am a little confused on the best way to do this.
>>
>> Regards
>>
>> Chris
>>
>> On Sat, May 15, 2010 at 12:18 PM, Madovsky <infos at madovsky.org> wrote:
>>
>>>  you can set it into your sip_profiles
>>>
>>> ----- Original Message -----
>>> *From:* Chris <freeswitch at gilligan.id.au>
>>> *To:* freeswitch-users at lists.freeswitch.org
>>> *Sent:* Friday, May 14, 2010 10:11 PM
>>> *Subject:* Re: [Freeswitch-users] mod_managed or a script,sending DTMF
>>> and receiving DTMF
>>>
>>> would you happen to know the default one in freeswitch? My system is for
>>> landlines anyway and that seems to work.  the voip phone was just for free
>>> testing so i did not have to pay for calls.
>>>
>>> On Sat, May 15, 2010 at 11:59 AM, Madovsky <infos at madovsky.org> wrote:
>>>
>>>>  read carefully RFC2833, RFC2976 and inband DTMF.
>>>> there are 3 major different ways to use DTMF. read the doc of your phone
>>>> to know which DTMF it uses
>>>>
>>>> F
>>>>
>>>>  ----- Original Message -----
>>>> *From:* Chris <freeswitch at gilligan.id.au>
>>>> *To:* freeswitch-users at lists.freeswitch.org
>>>>   *Sent:* Friday, May 14, 2010 9:45 PM
>>>> *Subject:* Re: [Freeswitch-users] mod_managed or a script,sending DTMF
>>>> and receiving DTMF
>>>>
>>>> ok worked out the problem. Thanks for the help.  my stupid softphone
>>>> does not support dtmf receiving ti seems.  used normal phone to call and it
>>>> all worked.  grrrr
>>>>
>>>> On Sat, May 15, 2010 at 11:39 AM, Madovsky <infos at madovsky.org> wrote:
>>>>
>>>>>  I use it with fs_cli
>>>>>
>>>>>   ----- Original Message -----
>>>>> *From:* Chris <freeswitch at gilligan.id.au>
>>>>> *To:* freeswitch-users at lists.freeswitch.org
>>>>> *Sent:* Friday, May 14, 2010 9:33 PM
>>>>> *Subject:* Re: [Freeswitch-users] mod_managed or a script,sending DTMF
>>>>> and receiving DTMF
>>>>>
>>>>> do you happen to be using ti from inside mod_managed or via a different
>>>>> means?  Any chance of a small sample?  maybe i am doing something stupid.
>>>>>
>>>>> On Sat, May 15, 2010 at 11:22 AM, Madovsky <infos at madovsky.org> wrote:
>>>>>
>>>>>>  it works, I use it everyday
>>>>>>
>>>>>>   ----- Original Message -----
>>>>>> *From:* Chris <freeswitch at gilligan.id.au>
>>>>>> *To:* freeswitch-users at lists.freeswitch.org
>>>>>> *Sent:* Friday, May 14, 2010 8:52 PM
>>>>>> *Subject:* Re: [Freeswitch-users] mod_managed or a script,sending
>>>>>> DTMF and receiving DTMF
>>>>>>
>>>>>> sadly uuid_send_dtmf did not seem to work
>>>>>>
>>>>>>
>>>>>> On Fri, May 14, 2010 at 11:10 AM, Chris <freeswitch at gilligan.id.au>wrote:
>>>>>>
>>>>>>> Thanks so that is the only way?  I will have to do some tests and see
>>>>>>> if i can capture dtmf at the same time as i send when i do it that way.  Any
>>>>>>> other options?
>>>>>>>
>>>>>>>
>>>>>>> On Thu, May 13, 2010 at 10:50 PM, Phillip Jones <
>>>>>>> pjintheusa at gmail.com> wrote:
>>>>>>>
>>>>>>>> Try using
>>>>>>>> http://wiki.freeswitch.org/wiki/Mod_commands#uuid_send_dtmf
>>>>>>>>
>>>>>>>>   On Thu, May 13, 2010 at 5:32 AM, Chris <freeswitch at gilligan.id.au
>>>>>>>> > wrote:
>>>>>>>>
>>>>>>>>>  HI,
>>>>>>>>> I am currently working on a project in mod_managed and i am trying
>>>>>>>>> to discover the best way to meet my requirements.  i am hoping someone will
>>>>>>>>> have some ideas.  This could be implemented in one of the other scripting
>>>>>>>>> language if needed.
>>>>>>>>>
>>>>>>>>> What i am trying to do is reprogram a remote device via the phone.
>>>>>>>>>  The device takes commands in the form of DTMF tones and responds in
>>>>>>>>> different DTMF tones depending on success or failure. An example of the flow
>>>>>>>>> would be
>>>>>>>>>
>>>>>>>>> freeswitch -> 342523#
>>>>>>>>> device -> 1
>>>>>>>>> freeswitch -> 356789#
>>>>>>>>> device ->2
>>>>>>>>>
>>>>>>>>> device always responds with one digit and freeswitch sends many.
>>>>>>>>>
>>>>>>>>> I was hoping to do this with a database and mod_managed but i can't
>>>>>>>>> workout how to send the DTMF in mod_managed unless i user audio files for it
>>>>>>>>> which seems to be the wrong way to go to me. While i would prefer a
>>>>>>>>> mod_managed solution i will take anything i can find.
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> Chris
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>  ------------------------------
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>  ------------------------------
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>  ------------------------------
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>  ------------------------------
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Tue May 18 16:45:25 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 18 May 2010 19:45:25 -0400
Subject: [Freeswitch-users] TLS on Windows
In-Reply-To: <746307.1047.qm@web33502.mail.mud.yahoo.com>
References: <746307.1047.qm@web33502.mail.mud.yahoo.com>
Message-ID: <49F5AAC2-8605-4AA3-A13A-B88D602906E3@jerris.com>

No one ever made the sofia build on windows work with openssl so it is not available until we get patches to add this.

Mike

On May 18, 2010, at 6:18 PM, Diego Toro wrote:

> I have the certificates, this isn't my problem.  Sofia says some like protocol is not supported.
> 
> Thank you
> 
> Diego Toro
> http://lacarretade.blogspot.com/
> 
> 
> --- On Tue, 5/18/10, leo.zibi at gmail.com <leo.zibi at gmail.com> wrote:
> 
>> From: leo.zibi at gmail.com <leo.zibi at gmail.com>
>> Subject: Re: [Freeswitch-users] TLS on Windows
>> To: freeswitch-users at lists.freeswitch.org
>> Date: Tuesday, May 18, 2010, 3:38 PM
>> Hi,
>> 
>> http://wiki.freeswitch.org/wiki/Generating_TLS_certificates_(win32)
>> 
>> -- 
>> Regards
>> P.
>> 
>> Diego Toro wrote:
>>> Greetings,
>>> 
>>> I have been interested about TLS on Windows and
>> FreeSWITCH, after web search and mailing list, I don't know
>> if is possible using FreeSWITCH on Windows to have security
>> communication with TLS. 
>>>   
>>> I made and installed OpenSSL and I have tried build
>> FreeSWITCH (libsofia_sip_ua_static) using OpenSSL, I created
>> the certificates and put it on conf/ssl directory, but the
>> internal profile faults when it's created, the sofia log
>> says some like protocol is not supported.
>>> 
>>> I have lastest git branch 1.0.head, windows 7 32
>> bits.
>>> 
>>> I appreciate any suggestion



From mike at jerris.com  Tue May 18 16:50:02 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 18 May 2010 19:50:02 -0400
Subject: [Freeswitch-users] SCTP for Windows
In-Reply-To: <9FFF1D7C542C46A2B98D5B8E787E7E4A@dell9400>
References: <527B81D4113C455A964AAE20C28C3D49@dell9400>
	<B80AF699-A323-4031-8900-3AD3081912CB@jerris.com>
	<20100518164235.3F93FD6247B00@bmail02.one.com>
	<9FFF1D7C542C46A2B98D5B8E787E7E4A@dell9400>
Message-ID: <21B967A9-3874-419E-9CFE-827DF2F6EBD9@jerris.com>

http://sourceforge.net/projects/sctpdrv/ seems to be the one.  It makes me think maybe its time to come up with combining a few of these offerings to have a single sctp api with a good license across multiple operating systems.  Does anyone know any of the key players in any of these open source stacks?

Mike

On May 18, 2010, at 4:07 PM, Jan Berger wrote:

> You need to download this from sourceforge so you get the english install, newest version and doc.
>  
> The source I looked at say this is a Cisco sctp stack ???
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jan Berger
> Sent: 18. mai 2010 18:43
> To: freeswitch-users at lists.freeswitch.org; freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] SCTP for Windows
>  
> Thanks mate,
>  
> Japanese ? Might actually work then :)
>  
> There is a German library as well, but it says GPL.
>  
> Jan 
> 
> On May 18, 2010 16:42 "Michael Jerris" <mike at jerris.com> wrote:
>> http://www.co-conv.jp/en/product/sctpDrv/20081224/
>>  
>> On May 18, 2010, at 8:09 AM, Jan Berger wrote:
>> 
>> 
>> Hi list,
>>  
>> Does anyone know about a proper SCTP library for Windows that is reliable and with a compatible licens?
>>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From moises.silva at gmail.com  Tue May 18 18:23:08 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Tue, 18 May 2010 21:23:08 -0400
Subject: [Freeswitch-users] span not defined error
In-Reply-To: <AANLkTikNAel0cKvf0h-8vO-bdhXU_WO_35kiWWWSC7BS@mail.gmail.com>
References: <AANLkTik9DQcT5fpgcfb6KqFUZT4vXUuAjqTE5pFLxW-R@mail.gmail.com>
	<AANLkTin9wP0f74qEyA5GjtpVo7YacbS2C6VE9VnO31wj@mail.gmail.com>
	<AANLkTikleSqsMITY0lNf-Qyeg5kOb2ICJFDs4gO3UeZO@mail.gmail.com>
	<AANLkTikd_xI3QFUeWjJfVzNs5nc1tuK9_WHEu46Jsekd@mail.gmail.com>
	<AANLkTinbeTgmLDAcxyF3B_JqO7J59XFTbTOtLio0ByaM@mail.gmail.com>
	<AANLkTim6OwUdrXQsdsSo9npyMVaezHlVLfhZfNGszEQP@mail.gmail.com>
	<AANLkTikNAel0cKvf0h-8vO-bdhXU_WO_35kiWWWSC7BS@mail.gmail.com>
Message-ID: <AANLkTinTzxRLu7nqfXvMQnntI5fZglDJ_2P2Q22Na8LP@mail.gmail.com>

Nice, glad it worked :-)

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com


On Tue, May 18, 2010 at 12:07 PM, Neil Patel <neilp at cs.stanford.edu> wrote:

> Hey Moises,
>
> Upgrading wanpipe and FS to the latest did the trick. Thanks!
>
> -Neil
>
> On Mon, May 17, 2010 at 7:44 AM, Moises Silva <moises.silva at gmail.com>wrote:
>
>> Try taking a protocol dump using wanpipemon on the d-channel so we can see
>> the full details of the call setup.
>>
>>
>> http://wiki.sangoma.com/wanpipe-wireshark-pcap-pri-bri-wan-t1-e1-tracing#pri_bri_wireshark
>>
>> Then paste a link here to the pcap file somewhere.
>>
>> Moises Silva
>> Senior Software Engineer
>> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
>> 9T3 Canada
>> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>>
>>
>> On Sun, May 16, 2010 at 8:16 PM, Neil Patel <neilp at cs.stanford.edu>wrote:
>>
>>> Thanks Moises, this is very helpful!
>>>
>>> After fixing the dialstring, I am still getting the following error from
>>> FS:
>>>
>>> > originate openzap/smg_prid/a/<num> &echo
>>> -ERR NORMAL_CIRCUIT_CONGESTION
>>>
>>> 2010-05-17 05:39:02.883264 [WARNING] ozmod_sangoma_boost.c:346 TX EVENT:
>>> CALL_START:(80) [w1g1] CSid=1 Seq=0 Cn=[FreeSWITCH] Cd=[9428500597]
>>> Ci=[0000000000] Rdnis=[]
>>> freeswitch at internal> 2010-05-17 05:39:02.955336 [WARNING]
>>> ozmod_sangoma_boost.c:1312 RX EVENT (N): CALL_START_NACK:(82) [w256g256]
>>> Rc=0 CSid=1 Seq=1
>>> 2010-05-17 05:39:02.955336 [WARNING] sangoma_boost_client.c:220 TX EVENT
>>> (N): CALL_START_NACK_ACK:(83) [w1g1] Rc=0 CSid=1 Seq=1
>>> 2010-05-17 05:39:02.956586 [ERR] mod_openzap.c:1154 No channels available
>>> 2010-05-17 05:39:02.956586 [ERR] switch_ivr_originate.c:2249 Cannot
>>> create outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION]
>>>
>>>
>>>
>>> Here is the relevant logging from /var/log/sangoma_mgd.log:
>>>
>>> May 17 05:39:02 otalo sangoma_prid: Rx SMG [CALL_START s:0 c:0 id:1]
>>> May 17 05:39:02 otalo sangoma_prid: g1:Outgoing call (Smg-ID:1)
>>> May 17 05:39:02 otalo sangoma_prid: s1:Outgoing call ChanRq:1
>>> Called-Nb[9428500597] Calling-Nb[0000000000] (Smg-ID:1)
>>> May 17 05:39:02 otalo sangoma_prid: Rx Tsoft [7:StatusIn id:0]
>>> May 17 05:39:02 otalo sangoma_prid: s1:Received cause-ind-An IE or
>>> parameter does not exist(99)
>>> May 17 05:39:02 otalo sangoma_prid: Rx Tsoft [2:DiscIn id:0]
>>> May 17 05:39:02 otalo sangoma_prid: s1c1:Remote released-Unknown(0)
>>> May 17 05:39:02 otalo sangoma_prid: Rx SMG [CALL_START_NACK_ACK s:0 c:0
>>> id:1]
>>> May 17 05:39:02 otalo sangoma_prid: g1:Call cleared (SMG-ID:1)
>>>
>>>
>>> I updated /etc/wanpipe/smg_pri.conf to have group=1 uncommented.
>>>
>>> Please note that my box is in India, making calls over a PRI line set up
>>> there. I also changed my tones.conf to match what's on the wiki<http://wiki.freeswitch.org/wiki/Tones.conf_Example>for India.
>>>
>>> Thanks,
>>> Neil
>>>
>>> On Sat, May 15, 2010 at 10:20 PM, Moises Silva <moises.silva at gmail.com>wrote:
>>>
>>>> Hi Again Neil,
>>>>
>>>> I just noticed your dial string is incorrect. The correct syntax is:
>>>>
>>>> OpenZAP/<span>/<chan_code>/[number]
>>>>
>>>> The span and chan code are mandatory. The number is optional ( FXS
>>>> channels do not require a number, they just ring the FXO device connected to
>>>> them).
>>>>
>>>> The span is either a number ( span id, the id is a number assigned in
>>>> the order in which the span is defined in openzap.conf ) or a span name also
>>>> as specified in the [span wanpipe <span_name>] line in openzap.conf
>>>>
>>>> The chan code is either a number ( for spans that support individual
>>>> channel selection, boost is NOT one of them ), or a channel hunting mode,
>>>> there is currently 2 modes, "a" is top down and "A" is bottom up.
>>>>
>>>> So, this is a valid string for you case:
>>>>
>>>> OpenZAP/smg_prid/a/<number>
>>>>
>>>> In the specific case of boost in socket mode ( openzap only supports
>>>> socket mode ) the number may contain @gX where X is a group ( for hunting as
>>>> configured in /etc/wanpipe/smg_pri.conf). Boost signaling are a special case
>>>> because the hunting for channels is not done within FreeSWITCH but in
>>>> sangoma_prid binary ( in the new OpenZAP version called FreeTDM this has
>>>> changed depending on configuration).
>>>>
>>>> Bottom line, this should work:
>>>>
>>>> OpenZAP/smg_prid/a/1234 at g1
>>>>
>>>> If you have g1 configured in /etc/wanpipe/smg_pri.conf
>>>>
>>>> Moises Silva
>>>> Senior Software Engineer
>>>> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
>>>> 9T3 Canada
>>>> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>>>>
>>>>
>>>> On Sat, May 15, 2010 at 6:26 PM, Neil Patel <neilp at cs.stanford.edu>wrote:
>>>>
>>>>> Span was originally in the boost section when I got this error, so I
>>>>> thought I'd try it in analog and both. None work.
>>>>>
>>>>> -Neil
>>>>>
>>>>> On Sat, May 15, 2010 at 3:18 PM, Moises Silva <moises.silva at gmail.com>wrote:
>>>>>
>>>>>> Why do you have 2 spans in openzap.conf.xml with the same name, in
>>>>>> both the boost and analog sections?
>>>>>> Moises Silva
>>>>>> Senior Software Engineer
>>>>>> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON
>>>>>> L3R 9T3 Canada
>>>>>> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>>>>>>
>>>>>>
>>>>>> On Sat, May 15, 2010 at 3:52 PM, Neil Patel <neilp at cs.stanford.edu>wrote:
>>>>>>
>>>>>>> Hi All,
>>>>>>>
>>>>>>> I am trying to dial out over my PRI, and am getting this error:
>>>>>>>
>>>>>>> 2010-05-16 01:01:21.452392 [CRIT] zap_io.c:1139 SPAN NOT DEFINED!
>>>>>>> 2010-05-16 01:01:21.452392 [ERR] mod_openzap.c:1154 No channels
>>>>>>> available
>>>>>>> 2010-05-16 01:01:21.452392 [ERR] switch_ivr_originate.c:2249 Cannot
>>>>>>> create outgoing channel of type [OpenZAP] cause: [NORMAL_CIRCUIT_CONGESTION]
>>>>>>>
>>>>>>>
>>>>>>> This is my openzap.conf:
>>>>>>>
>>>>>>> [span wanpipe smg_prid]
>>>>>>> name => smg_prid
>>>>>>> trunk_type =>e1
>>>>>>> b-channel => 1:1-15
>>>>>>> b-channel => 1:17-31
>>>>>>> trunk_type =>e1
>>>>>>> b-channel => 2:1-15
>>>>>>> b-channel => 2:17-31
>>>>>>>
>>>>>>>
>>>>>>> This is my openzap.conf.xml:
>>>>>>>
>>>>>>> <configuration name="openzap.conf" description="OpenZAP
>>>>>>> Configuration">
>>>>>>>   <settings>
>>>>>>>     <param name="debug" value="0"/>
>>>>>>>
>>>>>>>   </settings>
>>>>>>>   <boost_spans>
>>>>>>>         <span id="smg_prid">
>>>>>>>         <param name="dialplan" value="XML"/>
>>>>>>>         <param name="context" value="default"/>
>>>>>>>         </span>
>>>>>>>   </boost_spans>
>>>>>>>
>>>>>>>   <analog_spans>
>>>>>>>         <span id="smg_prid">
>>>>>>>         <param name="dialplan" value="XML"/>
>>>>>>>         <param name="context" value="default"/>
>>>>>>>
>>>>>>>         </span>
>>>>>>>   </analog_spans>
>>>>>>> </configuration>
>>>>>>>
>>>>>>>
>>>>>>> And here is the lua code I'm using to dial out:
>>>>>>>
>>>>>>> sessiondata = "OpenZAP/smg_prid/<num>"
>>>>>>> new_session = freeswitch.Session(sessiondata)
>>>>>>>
>>>>>>>
>>>>>>> What am I missing here?
>>>>>>> Thanks,
>>>>>>> Neil
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From neilp at cs.stanford.edu  Tue May 18 21:12:38 2010
From: neilp at cs.stanford.edu (Neil Patel)
Date: Tue, 18 May 2010 21:12:38 -0700
Subject: [Freeswitch-users] ESL: luamod not compiling
Message-ID: <AANLkTim4xu21uCbiNOgYliosNIuE1nINZgTQqW0XQEw4@mail.gmail.com>

I just downloaded today's snapshot and tried compiling luamod. I got the
error below:

root at server:/usr/src/freeswitch-snapshot/libs/esl# make luamod
make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x"
CFLAGS="-I/usr/src/freeswitch-snapshot/libs/esl/src/include -DHAVE_EDITLINE
-g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror
-Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes"
CXXFLAGS="-I/usr/src/freeswitch-snapshot/libs/esl/src/include
-DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
-Wno-unused-variable" CXX_CFLAGS="" -C lua
make[1]: Entering directory `/usr/src/freeswitch-snapshot/libs/esl/lua'
g++  -I/usr/src/freeswitch-snapshot/libs/esl/src/include -DHAVE_EDITLINE -g
-ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable
 -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o
esl_wrap.cpp:726:17: error: lua.h: No such file or directory
esl_wrap.cpp:727:21: error: lauxlib.h: No such file or directory
esl_wrap.cpp:745: error: ?lua_CFunction? does not name a type
esl_wrap.cpp:746: error: ?lua_CFunction? does not name a type
esl_wrap.cpp:761: error: ?lua_CFunction? does not name a type
esl_wrap.cpp:766: error: ?lua_CFunction? does not name a type
esl_wrap.cpp:767: error: ?lua_CFunction? does not name a type
esl_wrap.cpp:773: error: ?lua_CFunction? does not name a type
esl_wrap.cpp:850: error: ?lua_State? was not declared in this scope
esl_wrap.cpp:850: error: ?L? was not declared in this scope
esl_wrap.cpp:850: error: expected ?,? or ?;? before ?{? token
esl_wrap.cpp:861: error: variable or field ?SWIG_Lua_SetModule? declared
void
esl_wrap.cpp:861: error: ?lua_State? was not declared in this scope
esl_wrap.cpp:861: error: ?L? was not declared in this scope
esl_wrap.cpp:861: error: expected primary-expression before ?*? token
esl_wrap.cpp:861: error: ?module? was not declared in this scope
esl_wrap.cpp:875: error: ?lua_State? was not declared in this scope
esl_wrap.cpp:875: error: ?L? was not declared in this scope
esl_wrap.cpp:876: error: expected ?,? or ?;? before ?{? token
esl_wrap.cpp:887: error: ?lua_State? was not declared in this scope
esl_wrap.cpp:887: error: ?L? was not declared in this scope
esl_wrap.cpp:888: error: expected ?,? or ?;? before ?{? token
esl_wrap.cpp:3156: error: expected `}' at end of input
make[1]: *** [esl_wrap.o] Error 1
make[1]: Leaving directory `/usr/src/freeswitch-snapshot/libs/esl/lua'
make: *** [luamod] Error 2

I tried changing the includes for lua.h and lauxlib.h to a place I had them
(/usr/include/lua5.1/), but then I got stuck on this error:

root at server:/usr/src/freeswitch-snapshot/libs/esl# make luamod
make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x"
CFLAGS="-I/usr/src/freeswitch-snapshot/libs/esl/src/include -DHAVE_EDITLINE
-g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror
-Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes"
CXXFLAGS="-I/usr/src/freeswitch-snapshot/libs/esl/src/include
-DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
-Wno-unused-variable" CXX_CFLAGS="" -C lua
make[1]: Entering directory `/usr/src/freeswitch-snapshot/libs/esl/lua'
g++  -I/usr/src/freeswitch-snapshot/libs/esl/src/include -DHAVE_EDITLINE -g
-ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable
 -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o
g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -llua -lpthread -o ESL.so
-L.
/usr/bin/ld: cannot find -llua
collect2: ld returned 1 exit status
make[1]: *** [ESL.so] Error 1
make[1]: Leaving directory `/usr/src/freeswitch-snapshot/libs/esl/lua'
make: *** [luamod] Error 2

I couldn't quite make sense of this. Thanks in advance.
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From brian at freeswitch.org  Tue May 18 21:23:38 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 18 May 2010 23:23:38 -0500
Subject: [Freeswitch-users] ESL: luamod not compiling
In-Reply-To: <AANLkTim4xu21uCbiNOgYliosNIuE1nINZgTQqW0XQEw4@mail.gmail.com>
References: <AANLkTim4xu21uCbiNOgYliosNIuE1nINZgTQqW0XQEw4@mail.gmail.com>
Message-ID: <ACB6B6B0-2BA1-4E3C-9B5D-6D59728DBAB5@freeswitch.org>

you ace to manually compile lua due to it being compiled without the -fPIC flag.

/b

Sent from my iPad

On May 18, 2010, at 11:12 PM, Neil Patel <neilp at cs.stanford.edu> wrote:

> I just downloaded today's snapshot and tried compiling luamod. I got the error below:
> 
> root at server:/usr/src/freeswitch-snapshot/libs/esl# make luamod
> make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-I/usr/src/freeswitch-snapshot/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes" CXXFLAGS="-I/usr/src/freeswitch-snapshot/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" CXX_CFLAGS="" -C lua
> make[1]: Entering directory `/usr/src/freeswitch-snapshot/libs/esl/lua'
> g++  -I/usr/src/freeswitch-snapshot/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable  -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o
> esl_wrap.cpp:726:17: error: lua.h: No such file or directory
> esl_wrap.cpp:727:21: error: lauxlib.h: No such file or directory
> esl_wrap.cpp:745: error: ?lua_CFunction? does not name a type
> esl_wrap.cpp:746: error: ?lua_CFunction? does not name a type
> esl_wrap.cpp:761: error: ?lua_CFunction? does not name a type
> esl_wrap.cpp:766: error: ?lua_CFunction? does not name a type
> esl_wrap.cpp:767: error: ?lua_CFunction? does not name a type
> esl_wrap.cpp:773: error: ?lua_CFunction? does not name a type
> esl_wrap.cpp:850: error: ?lua_State? was not declared in this scope
> esl_wrap.cpp:850: error: ?L? was not declared in this scope
> esl_wrap.cpp:850: error: expected ?,? or ?;? before ?{? token
> esl_wrap.cpp:861: error: variable or field ?SWIG_Lua_SetModule? declared void
> esl_wrap.cpp:861: error: ?lua_State? was not declared in this scope
> esl_wrap.cpp:861: error: ?L? was not declared in this scope
> esl_wrap.cpp:861: error: expected primary-expression before ?*? token
> esl_wrap.cpp:861: error: ?module? was not declared in this scope
> esl_wrap.cpp:875: error: ?lua_State? was not declared in this scope
> esl_wrap.cpp:875: error: ?L? was not declared in this scope
> esl_wrap.cpp:876: error: expected ?,? or ?;? before ?{? token
> esl_wrap.cpp:887: error: ?lua_State? was not declared in this scope
> esl_wrap.cpp:887: error: ?L? was not declared in this scope
> esl_wrap.cpp:888: error: expected ?,? or ?;? before ?{? token
> esl_wrap.cpp:3156: error: expected `}' at end of input
> make[1]: *** [esl_wrap.o] Error 1
> make[1]: Leaving directory `/usr/src/freeswitch-snapshot/libs/esl/lua'
> make: *** [luamod] Error 2
> 
> I tried changing the includes for lua.h and lauxlib.h to a place I had them (/usr/include/lua5.1/), but then I got stuck on this error:
> 
> root at server:/usr/src/freeswitch-snapshot/libs/esl# make luamod
> make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-I/usr/src/freeswitch-snapshot/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes" CXXFLAGS="-I/usr/src/freeswitch-snapshot/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" CXX_CFLAGS="" -C lua
> make[1]: Entering directory `/usr/src/freeswitch-snapshot/libs/esl/lua'
> g++  -I/usr/src/freeswitch-snapshot/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable  -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o
> g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -llua -lpthread -o ESL.so -L. 
> /usr/bin/ld: cannot find -llua
> collect2: ld returned 1 exit status
> make[1]: *** [ESL.so] Error 1
> make[1]: Leaving directory `/usr/src/freeswitch-snapshot/libs/esl/lua'
> make: *** [luamod] Error 2
> 
> I couldn't quite make sense of this. Thanks in advance.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From afc007 at gmail.com  Tue May 18 15:22:16 2010
From: afc007 at gmail.com (Alexander Caskey)
Date: Tue, 18 May 2010 15:22:16 -0700
Subject: [Freeswitch-users] how can I use FS to record a conversation
	(inbound and outbound) in Windows environment (.wav preferred)
Message-ID: <AANLkTilGX7tu46-CLyYTKFfP6rcDrvPikUaxcnheLs3U@mail.gmail.com>

(With advance apologies for asking a beginner question)

I want FS to record both sides of a conversation and save the result as a
single .wav file (Windows environment)

I have grepped through the wiki and this list but haven't been able to find
the answer I'm looking (with example config file entries etc).

Any help would be much appreciated.

Alec
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From mike at jerris.com  Tue May 18 21:26:19 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 19 May 2010 00:26:19 -0400
Subject: [Freeswitch-users] ESL: luamod not compiling
In-Reply-To: <AANLkTim4xu21uCbiNOgYliosNIuE1nINZgTQqW0XQEw4@mail.gmail.com>
References: <AANLkTim4xu21uCbiNOgYliosNIuE1nINZgTQqW0XQEw4@mail.gmail.com>
Message-ID: <0F8DE493-DC70-44D4-909C-D8F74EF92840@jerris.com>


On May 19, 2010, at 12:12 AM, Neil Patel wrote:

> I just downloaded today's snapshot and tried compiling luamod. I got the error below:
> 
> root at server:/usr/src/freeswitch-snapshot/libs/esl# make luamod
> make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-I/usr/src/freeswitch-snapshot/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes" CXXFLAGS="-I/usr/src/freeswitch-snapshot/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" CXX_CFLAGS="" -C lua
> make[1]: Entering directory `/usr/src/freeswitch-snapshot/libs/esl/lua'
> g++  -I/usr/src/freeswitch-snapshot/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable  -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o
> esl_wrap.cpp:726:17: error: lua.h: No such file or directory
> esl_wrap.cpp:727:21: error: lauxlib.h: No such file or directory
> esl_wrap.cpp:745: error: ?lua_CFunction? does not name a type
> esl_wrap.cpp:746: error: ?lua_CFunction? does not name a type
> esl_wrap.cpp:761: error: ?lua_CFunction? does not name a type
> esl_wrap.cpp:766: error: ?lua_CFunction? does not name a type
> esl_wrap.cpp:767: error: ?lua_CFunction? does not name a type
> esl_wrap.cpp:773: error: ?lua_CFunction? does not name a type
> esl_wrap.cpp:850: error: ?lua_State? was not declared in this scope
> esl_wrap.cpp:850: error: ?L? was not declared in this scope
> esl_wrap.cpp:850: error: expected ?,? or ?;? before ?{? token
> esl_wrap.cpp:861: error: variable or field ?SWIG_Lua_SetModule? declared void
> esl_wrap.cpp:861: error: ?lua_State? was not declared in this scope
> esl_wrap.cpp:861: error: ?L? was not declared in this scope
> esl_wrap.cpp:861: error: expected primary-expression before ?*? token
> esl_wrap.cpp:861: error: ?module? was not declared in this scope
> esl_wrap.cpp:875: error: ?lua_State? was not declared in this scope
> esl_wrap.cpp:875: error: ?L? was not declared in this scope
> esl_wrap.cpp:876: error: expected ?,? or ?;? before ?{? token
> esl_wrap.cpp:887: error: ?lua_State? was not declared in this scope
> esl_wrap.cpp:887: error: ?L? was not declared in this scope
> esl_wrap.cpp:888: error: expected ?,? or ?;? before ?{? token
> esl_wrap.cpp:3156: error: expected `}' at end of input
> make[1]: *** [esl_wrap.o] Error 1
> make[1]: Leaving directory `/usr/src/freeswitch-snapshot/libs/esl/lua'
> make: *** [luamod] Error 2
> 
> I tried changing the includes for lua.h and lauxlib.h to a place I had them (/usr/include/lua5.1/), but then I got stuck on this error:
> 
> root at server:/usr/src/freeswitch-snapshot/libs/esl# make luamod
> make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-I/usr/src/freeswitch-snapshot/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes" CXXFLAGS="-I/usr/src/freeswitch-snapshot/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" CXX_CFLAGS="" -C lua
> make[1]: Entering directory `/usr/src/freeswitch-snapshot/libs/esl/lua'
> g++  -I/usr/src/freeswitch-snapshot/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable  -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o
> g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -llua -lpthread -o ESL.so -L. 
> /usr/bin/ld: cannot find -llua
> collect2: ld returned 1 exit status
> make[1]: *** [ESL.so] Error 1
> make[1]: Leaving directory `/usr/src/freeswitch-snapshot/libs/esl/lua'
> make: *** [luamod] Error 2
> 
> I couldn't quite make sense of this. Thanks in advance.


It means it can't find liblua.so or liblua.a in the lib search path or in any -L arg passed in LDFLAGS



From mike at jerris.com  Tue May 18 21:29:06 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 19 May 2010 00:29:06 -0400
Subject: [Freeswitch-users] FS on Sparc based Architecture
In-Reply-To: <AANLkTiklyJ-DwEyRxcEwd2mpvDqrBucZANCR3U2bGuFh@mail.gmail.com>
References: <AANLkTiklyJ-DwEyRxcEwd2mpvDqrBucZANCR3U2bGuFh@mail.gmail.com>
Message-ID: <A30B4BDA-B88D-4B57-B77D-E6E78DD291B9@jerris.com>

It should work fine, yes.

On May 18, 2010, at 4:14 PM, RR wrote:

> Hello List,
> 
> I did a bit of hunting here and there but couldn't find anything that mentions any one having any experience/success/failure with compiling/building FS under Solaris (other than http://wiki.freeswitch.org/wiki/Installation_Guide#Solaris) running over a Sparc based architecture. For all I know, it may have nothing to do with it being Solaris on Sparc or Intel but does anyone know and can verify for sure that using the instructions on the Wiki, FS can be built for Solaris on Sparc and will/does stable and seamlessly as it does on Linux etc?
> 
> Sorry if this is a silly question but just needed to ask for my own sanity :)
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From neilp at cs.stanford.edu  Tue May 18 21:33:07 2010
From: neilp at cs.stanford.edu (Neil Patel)
Date: Tue, 18 May 2010 21:33:07 -0700
Subject: [Freeswitch-users] ESL: luamod not compiling
In-Reply-To: <0F8DE493-DC70-44D4-909C-D8F74EF92840@jerris.com>
References: <AANLkTim4xu21uCbiNOgYliosNIuE1nINZgTQqW0XQEw4@mail.gmail.com> 
	<0F8DE493-DC70-44D4-909C-D8F74EF92840@jerris.com>
Message-ID: <AANLkTilExUZf6QgWiJ_g3ZqMNdJ-uvrAtcZPsoIXDduq@mail.gmail.com>

On Tue, May 18, 2010 at 9:26 PM, Michael Jerris <mike at jerris.com> wrote:

>
> On May 19, 2010, at 12:12 AM, Neil Patel wrote:
>
> > I just downloaded today's snapshot and tried compiling luamod. I got the
> error below:
> >
> > root at server:/usr/src/freeswitch-snapshot/libs/esl# make luamod
> > make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x"
> CFLAGS="-I/usr/src/freeswitch-snapshot/libs/esl/src/include -DHAVE_EDITLINE
> -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror
> -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes"
> CXXFLAGS="-I/usr/src/freeswitch-snapshot/libs/esl/src/include
> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
> -Wno-unused-variable" CXX_CFLAGS="" -C lua
> > make[1]: Entering directory `/usr/src/freeswitch-snapshot/libs/esl/lua'
> > g++  -I/usr/src/freeswitch-snapshot/libs/esl/src/include -DHAVE_EDITLINE
> -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable
>  -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o
> > esl_wrap.cpp:726:17: error: lua.h: No such file or directory
> > esl_wrap.cpp:727:21: error: lauxlib.h: No such file or directory
> > esl_wrap.cpp:745: error: ?lua_CFunction? does not name a type
> > esl_wrap.cpp:746: error: ?lua_CFunction? does not name a type
> > esl_wrap.cpp:761: error: ?lua_CFunction? does not name a type
> > esl_wrap.cpp:766: error: ?lua_CFunction? does not name a type
> > esl_wrap.cpp:767: error: ?lua_CFunction? does not name a type
> > esl_wrap.cpp:773: error: ?lua_CFunction? does not name a type
> > esl_wrap.cpp:850: error: ?lua_State? was not declared in this scope
> > esl_wrap.cpp:850: error: ?L? was not declared in this scope
> > esl_wrap.cpp:850: error: expected ?,? or ?;? before ?{? token
> > esl_wrap.cpp:861: error: variable or field ?SWIG_Lua_SetModule? declared
> void
> > esl_wrap.cpp:861: error: ?lua_State? was not declared in this scope
> > esl_wrap.cpp:861: error: ?L? was not declared in this scope
> > esl_wrap.cpp:861: error: expected primary-expression before ?*? token
> > esl_wrap.cpp:861: error: ?module? was not declared in this scope
> > esl_wrap.cpp:875: error: ?lua_State? was not declared in this scope
> > esl_wrap.cpp:875: error: ?L? was not declared in this scope
> > esl_wrap.cpp:876: error: expected ?,? or ?;? before ?{? token
> > esl_wrap.cpp:887: error: ?lua_State? was not declared in this scope
> > esl_wrap.cpp:887: error: ?L? was not declared in this scope
> > esl_wrap.cpp:888: error: expected ?,? or ?;? before ?{? token
> > esl_wrap.cpp:3156: error: expected `}' at end of input
> > make[1]: *** [esl_wrap.o] Error 1
> > make[1]: Leaving directory `/usr/src/freeswitch-snapshot/libs/esl/lua'
> > make: *** [luamod] Error 2
> >
> > I tried changing the includes for lua.h and lauxlib.h to a place I had
> them (/usr/include/lua5.1/), but then I got stuck on this error:
> >
> > root at server:/usr/src/freeswitch-snapshot/libs/esl# make luamod
> > make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x"
> CFLAGS="-I/usr/src/freeswitch-snapshot/libs/esl/src/include -DHAVE_EDITLINE
> -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror
> -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes"
> CXXFLAGS="-I/usr/src/freeswitch-snapshot/libs/esl/src/include
> -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror
> -Wno-unused-variable" CXX_CFLAGS="" -C lua
> > make[1]: Entering directory `/usr/src/freeswitch-snapshot/libs/esl/lua'
> > g++  -I/usr/src/freeswitch-snapshot/libs/esl/src/include -DHAVE_EDITLINE
> -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable
>  -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o
> > g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -llua -lpthread -o ESL.so
> -L.
> > /usr/bin/ld: cannot find -llua
> > collect2: ld returned 1 exit status
> > make[1]: *** [ESL.so] Error 1
> > make[1]: Leaving directory `/usr/src/freeswitch-snapshot/libs/esl/lua'
> > make: *** [luamod] Error 2
> >
> > I couldn't quite make sense of this. Thanks in advance.
>
>
> It means it can't find liblua.so or liblua.a in the lib search path or in
> any -L arg passed in LDFLAGS
>

I don't have liblua.so. liblua.a is here:
freeswitch/src/mod/languages/mod_lua/lua/liblua.a

Is this why I have to compile lua manually? Should it be with or without
-fPIC?


>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Tue May 18 22:09:04 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 19 May 2010 00:09:04 -0500
Subject: [Freeswitch-users] ESL: luamod not compiling
In-Reply-To: <AANLkTilExUZf6QgWiJ_g3ZqMNdJ-uvrAtcZPsoIXDduq@mail.gmail.com>
References: <AANLkTim4xu21uCbiNOgYliosNIuE1nINZgTQqW0XQEw4@mail.gmail.com>
	<0F8DE493-DC70-44D4-909C-D8F74EF92840@jerris.com>
	<AANLkTilExUZf6QgWiJ_g3ZqMNdJ-uvrAtcZPsoIXDduq@mail.gmail.com>
Message-ID: <CA3F5374-1963-4FDC-96B2-FACF8824A5A4@freeswitch.org>

always with -fPIC 

Sent from my iPad

On May 18, 2010, at 11:33 PM, Neil Patel <neilp at cs.stanford.edu> wrote:

> Is this why I have to compile lua manually? Should it be with or without -fPIC?


From rupa at rupa.com  Tue May 18 22:12:30 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 19 May 2010 00:12:30 -0500
Subject: [Freeswitch-users] how can I use FS to record a conversation
	(inbound and outbound) in Windows environment (.wav preferred)
In-Reply-To: <AANLkTilGX7tu46-CLyYTKFfP6rcDrvPikUaxcnheLs3U@mail.gmail.com>
References: <AANLkTilGX7tu46-CLyYTKFfP6rcDrvPikUaxcnheLs3U@mail.gmail.com>
Message-ID: <AANLkTilZXANq7ZA6M2liZLxqI8YpV2M4GYCyBOvV22tV@mail.gmail.com>

Wiki has an example:

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

On Tue, May 18, 2010 at 5:22 PM, Alexander Caskey <afc007 at gmail.com> wrote:

> (With advance apologies for asking a beginner question)
>
> I want FS to record both sides of a conversation and save the result as a
> single .wav file (Windows environment)
>
> I have grepped through the wiki and this list but haven't been able to find
> the answer I'm looking (with example config file entries etc).
>
> Any help would be much appreciated.
>
> Alec
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
-Rupa
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From jor3l at foravatars.com  Tue May 18 23:25:28 2010
From: jor3l at foravatars.com (Jor3l Boa)
Date: Wed, 19 May 2010 01:25:28 -0500
Subject: [Freeswitch-users] Some help needed with Sofia
Message-ID: <AANLkTin0TCZYqIxmPDWouisxckz832kSGGRm8yRyJRtB@mail.gmail.com>

Hello there,

I'm trying to run FreeSwitch and OpenSim.. Since I'm new with FS, it is
pretty hard to get figure out something and solve it.. so far, I'm able to
register an user and connect to FS but I'm getting this everytime the user
is connected:

2010-05-19 01:10:57.179287 [NOTICE] switch_channel.c:675 New Channel
sofia/internal/xdPCKuP4mQdqNMLD4z3su0w==@192.168.0.171[4a0251c0-630d-11df-96ad-d1a299ca17c4]
2010-05-19 01:10:57.188530 [INFO] mod_dialplan_xml.c:331 Processing
UserName->conf-xNTc2OWU3NWUtODg0My00NmM5LTliMjktOWM5MWJhNmJjNjVl in context
public
2010-05-19 01:10:57.190542 [NOTICE] switch_core_state_machine.c:185
sofia/internal/xdPCKuP4mQdqNMLD4z3su0w==@192.168.0.171 has executed the last
dialplan instruction, hanging up.
2010-05-19 01:10:57.190542 [NOTICE] switch_core_state_machine.c:187 Hangup
sofia/internal/xdPCKuP4mQdqNMLD4z3su0w==@192.168.0.171 [CS_EXECUTE]
[NORMAL_CLEARING]
2010-05-19 01:10:57.200551 [NOTICE] switch_core_session.c:1188 Session 1765
(sofia/internal/xdPCKuP4mQdqNMLD4z3su0w==@192.168.0.171) Ended
2010-05-19 01:10:57.200551 [NOTICE] switch_core_session.c:1190 Close Channel
sofia/internal/xdPCKuP4mQdqNMLD4z3su0w==@192.168.0.171 [CS_DESTROY]

This repeats every second (New channel -> Process user in context public ->
hanging up -> close),

Is there something needed to allow anyone to use FS? (since anyone is able
to connect to OpenSim and use the voice service),
Why it is hanging up the call?

Thanks and sorry for my english

Regards,
Jorel
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From peter.olsson at visionutveckling.se  Tue May 18 23:37:55 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 19 May 2010 08:37:55 +0200
Subject: [Freeswitch-users] TLS on Windows
In-Reply-To: <49F5AAC2-8605-4AA3-A13A-B88D602906E3@jerris.com>
References: <746307.1047.qm@web33502.mail.mud.yahoo.com>
	<49F5AAC2-8605-4AA3-A13A-B88D602906E3@jerris.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C557D44E0CD@cooper>

Someone tried to do a patch once (FSBUILD-70), but I don't think it was ever accepted. It would probably need some more work.

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Michael Jerris
Skickat: den 19 maj 2010 01:45
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] TLS on Windows

No one ever made the sofia build on windows work with openssl so it is not available until we get patches to add this.

Mike

On May 18, 2010, at 6:18 PM, Diego Toro wrote:

> I have the certificates, this isn't my problem.  Sofia says some like protocol is not supported.
> 
> Thank you
> 
> Diego Toro
> http://lacarretade.blogspot.com/
> 
> 
> --- On Tue, 5/18/10, leo.zibi at gmail.com <leo.zibi at gmail.com> wrote:
> 
>> From: leo.zibi at gmail.com <leo.zibi at gmail.com>
>> Subject: Re: [Freeswitch-users] TLS on Windows
>> To: freeswitch-users at lists.freeswitch.org
>> Date: Tuesday, May 18, 2010, 3:38 PM
>> Hi,
>> 
>> http://wiki.freeswitch.org/wiki/Generating_TLS_certificates_(win32)
>> 
>> -- 
>> Regards
>> P.
>> 
>> Diego Toro wrote:
>>> Greetings,
>>> 
>>> I have been interested about TLS on Windows and
>> FreeSWITCH, after web search and mailing list, I don't know
>> if is possible using FreeSWITCH on Windows to have security
>> communication with TLS. 
>>>   
>>> I made and installed OpenSSL and I have tried build
>> FreeSWITCH (libsofia_sip_ua_static) using OpenSSL, I created
>> the certificates and put it on conf/ssl directory, but the
>> internal profile faults when it's created, the sofia log
>> says some like protocol is not supported.
>>> 
>>> I have lastest git branch 1.0.head, windows 7 32
>> bits.
>>> 
>>> I appreciate any suggestion


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

!DSPAM:4bf3288332932133842177!



From babak.freeswitch at gmail.com  Tue May 18 23:56:26 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Wed, 19 May 2010 11:26:26 +0430
Subject: [Freeswitch-users] mod_managed playback
Message-ID: <AANLkTinaORyErIYzADk7yg243VJEMjhUP4Vt7tuVbQxj@mail.gmail.com>

Hi
what is the difference between these tow in mod_managed???
Session.Execute("playback",path);
Session.StreamFile(path,-1);
because when I use the first one every thing is ok, but second one skips 3
or 4 sec of the file beginning.
thanks
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From jan.berger at video24.no  Wed May 19 00:50:04 2010
From: jan.berger at video24.no (Jan Berger)
Date: Wed, 19 May 2010 09:50:04 +0200
Subject: [Freeswitch-users] SCTP for Windows
In-Reply-To: <21B967A9-3874-419E-9CFE-827DF2F6EBD9@jerris.com>
References: <527B81D4113C455A964AAE20C28C3D49@dell9400>
	<B80AF699-A323-4031-8900-3AD3081912CB@jerris.com>
	<20100518164235.3F93FD6247B00@bmail02.one.com>
	<9FFF1D7C542C46A2B98D5B8E787E7E4A@dell9400>
	<21B967A9-3874-419E-9CFE-827DF2F6EBD9@jerris.com>
Message-ID: <E3050236FAB6417A9755EE35CA7271B5@dell9400>

Mike, 

 

The key players should be the operating systems. SCTP is included on
Sun/Solaris and Linux kernel - it's Windows that don't have it. Using sctp
should be very simple - a single parameter change on tcp/ip usage. 

 

---

 

This one don't install properly yet - WSCInstallProvider=11003 - the
services decline to start, but I will see what I can achieve. 

 

If you download this you find what seems to b a Cisco stack with bsd license
inside, and it seems to be maintained. The sctp code looks very clean and is
BSD license. I do find hint's on Cisco's pages that their sctp stack is open
source, but I have not found a direct link to it except for this.

 

---

 

The alternative for Windows is this http://www.sctp.be/sctplib/index.htm.
but this is GPL and not maintained.

 

Jan

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Jerris
Sent: 19. mai 2010 01:50
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] SCTP for Windows

 

http://sourceforge.net/projects/sctpdrv/ seems to be the one.  It makes me
think maybe its time to come up with combining a few of these offerings to
have a single sctp api with a good license across multiple operating
systems.  Does anyone know any of the key players in any of these open
source stacks?

 

Mike

 

On May 18, 2010, at 4:07 PM, Jan Berger wrote:





You need to download this from sourceforge so you get the english install,
newest version and doc.

 

The source I looked at say this is a Cisco sctp stack ???

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jan
Berger
Sent: 18. mai 2010 18:43
To: freeswitch-users at lists.freeswitch.org;
freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] SCTP for Windows

 

Thanks mate,

 

Japanese ? Might actually work then :)

 

There is a German library as well, but it says GPL. 

 

Jan 

On May 18, 2010 16:42 "Michael Jerris"  <mailto:mike at jerris.com>
<mike at jerris.com> wrote:

http://www.co-conv.jp/en/product/sctpDrv/20081224/ 

 

On May 18, 2010, at 8:09 AM, Jan Berger wrote:






Hi list,

 

Does anyone know about a proper SCTP library for Windows that is reliable
and with a compatible licens?

 

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



 

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From babak.freeswitch at gmail.com  Wed May 19 02:35:07 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Wed, 19 May 2010 14:05:07 +0430
Subject: [Freeswitch-users] mod_managed bind problem
Message-ID: <AANLkTilO75HysCaRkSu6oOmorPsItLhojiEjjRqpqdPW@mail.gmail.com>

Hi
after updating to latest git today I can not bind on events anymore:
EventConsumer con = new EventConsumer("all", null);

2010-05-19 14:04:32.984375 [ERR] switch_cpp.cpp:81 Cannot bind to ALL
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From mike at jerris.com  Wed May 19 06:40:32 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 19 May 2010 09:40:32 -0400
Subject: [Freeswitch-users] Some help needed with Sofia
In-Reply-To: <AANLkTin0TCZYqIxmPDWouisxckz832kSGGRm8yRyJRtB@mail.gmail.com>
References: <AANLkTin0TCZYqIxmPDWouisxckz832kSGGRm8yRyJRtB@mail.gmail.com>
Message-ID: <4FA51EFF-25E2-48F5-A927-64C56919B268@jerris.com>

This call isn't matching any extensions in the dialplan, or there is no actions in the dialplan for it to execute.

Mike

On May 19, 2010, at 2:25 AM, Jor3l Boa wrote:

> Hello there, 
> 
> I'm trying to run FreeSwitch and OpenSim.. Since I'm new with FS, it is pretty hard to get figure out something and solve it.. so far, I'm able to register an user and connect to FS but I'm getting this everytime the user is connected:
> 
> 2010-05-19 01:10:57.179287 [NOTICE] switch_channel.c:675 New Channel sofia/internal/xdPCKuP4mQdqNMLD4z3su0w==@192.168.0.171 [4a0251c0-630d-11df-96ad-d1a299ca17c4]
> 2010-05-19 01:10:57.188530 [INFO] mod_dialplan_xml.c:331 Processing UserName->conf-xNTc2OWU3NWUtODg0My00NmM5LTliMjktOWM5MWJhNmJjNjVl in context public
> 2010-05-19 01:10:57.190542 [NOTICE] switch_core_state_machine.c:185 sofia/internal/xdPCKuP4mQdqNMLD4z3su0w==@192.168.0.171 has executed the last dialplan instruction, hanging up.
> 2010-05-19 01:10:57.190542 [NOTICE] switch_core_state_machine.c:187 Hangup sofia/internal/xdPCKuP4mQdqNMLD4z3su0w==@192.168.0.171 [CS_EXECUTE] [NORMAL_CLEARING]
> 2010-05-19 01:10:57.200551 [NOTICE] switch_core_session.c:1188 Session 1765 (sofia/internal/xdPCKuP4mQdqNMLD4z3su0w==@192.168.0.171) Ended
> 2010-05-19 01:10:57.200551 [NOTICE] switch_core_session.c:1190 Close Channel sofia/internal/xdPCKuP4mQdqNMLD4z3su0w==@192.168.0.171 [CS_DESTROY]
> 
> This repeats every second (New channel -> Process user in context public -> hanging up -> close), 
> 
> Is there something needed to allow anyone to use FS? (since anyone is able to connect to OpenSim and use the voice service),
> Why it is hanging up the call?
> 
> Thanks and sorry for my english 
> 
> Regards,
> Jorel
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Wed May 19 06:41:44 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 19 May 2010 09:41:44 -0400
Subject: [Freeswitch-users] TLS on Windows
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C557D44E0CD@cooper>
References: <746307.1047.qm@web33502.mail.mud.yahoo.com>
	<49F5AAC2-8605-4AA3-A13A-B88D602906E3@jerris.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C557D44E0CD@cooper>
Message-ID: <0B49099E-83DA-4324-827C-E93F52A05FBB@jerris.com>

That patch used static compiled openssl.  This means the openssl could be built against a different version of the c runtime, which can cause issues.  An acceptable patch would have to build openssl as well.

Mike

On May 19, 2010, at 2:37 AM, Peter Olsson wrote:

> Someone tried to do a patch once (FSBUILD-70), but I don't think it was ever accepted. It would probably need some more work.
> 
> /Peter
> 
> 
> -----Ursprungligt meddelande-----
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Michael Jerris
> Skickat: den 19 maj 2010 01:45
> Till: freeswitch-users at lists.freeswitch.org
> ?mne: Re: [Freeswitch-users] TLS on Windows
> 
> No one ever made the sofia build on windows work with openssl so it is not available until we get patches to add this.
> 
> Mike
> 
> On May 18, 2010, at 6:18 PM, Diego Toro wrote:
> 
>> I have the certificates, this isn't my problem.  Sofia says some like protocol is not supported.
>> 
>> Thank you
>> 
>> Diego Toro
>> http://lacarretade.blogspot.com/
>> 
>> 
>> --- On Tue, 5/18/10, leo.zibi at gmail.com <leo.zibi at gmail.com> wrote:
>> 
>>> From: leo.zibi at gmail.com <leo.zibi at gmail.com>
>>> Subject: Re: [Freeswitch-users] TLS on Windows
>>> To: freeswitch-users at lists.freeswitch.org
>>> Date: Tuesday, May 18, 2010, 3:38 PM
>>> Hi,
>>> 
>>> http://wiki.freeswitch.org/wiki/Generating_TLS_certificates_(win32)
>>> 
>>> -- 
>>> Regards
>>> P.
>>> 
>>> Diego Toro wrote:
>>>> Greetings,
>>>> 
>>>> I have been interested about TLS on Windows and
>>> FreeSWITCH, after web search and mailing list, I don't know
>>> if is possible using FreeSWITCH on Windows to have security
>>> communication with TLS. 
>>>> 
>>>> I made and installed OpenSSL and I have tried build
>>> FreeSWITCH (libsofia_sip_ua_static) using OpenSSL, I created
>>> the certificates and put it on conf/ssl directory, but the
>>> internal profile faults when it's created, the sofia log
>>> says some like protocol is not supported.
>>>> 
>>>> I have lastest git branch 1.0.head, windows 7 32
>>> bits.
>>>> 
>>>> I appreciate any suggestion
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> !DSPAM:4bf3288332932133842177!
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From davis.erwin at gmail.com  Wed May 19 07:21:16 2010
From: davis.erwin at gmail.com (Erwin Davis)
Date: Wed, 19 May 2010 10:21:16 -0400
Subject: [Freeswitch-users] curl_xml
Message-ID: <AANLkTiltmx-qzAAgkQjUcCAGsAMp9_X573zbQic24pf7@mail.gmail.com>

Hi, I am trying to use curl_xml to dynamically send back the response for
directory query from a webserver.
But when the FS received the response from the web server, it kept sending
403 forbidden response to the sip client (x-lite).
I post the FS log from  the console in
http://pastebin.freeswitch.org/13007and the XML response log in
http://pastebin.freeswitch.org/13007. Thanks for your help,
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From brian at freeswitch.org  Wed May 19 07:25:39 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 19 May 2010 09:25:39 -0500
Subject: [Freeswitch-users] curl_xml
In-Reply-To: <AANLkTiltmx-qzAAgkQjUcCAGsAMp9_X573zbQic24pf7@mail.gmail.com>
References: <AANLkTiltmx-qzAAgkQjUcCAGsAMp9_X573zbQic24pf7@mail.gmail.com>
Message-ID: <DF566DB4-AA3D-468C-A746-93F57BFED1D3@freeswitch.org>

What exactly are you returning to freeswitch?  What does the resulting XML look like? 

/b

On May 19, 2010, at 9:21 AM, Erwin Davis wrote:

> Hi, I am trying to use curl_xml to dynamically send back the response for directory query from a webserver.
> But when the FS received the response from the web server, it kept sending 403 forbidden response to the sip client (x-lite).
> I post the FS log from  the console in http://pastebin.freeswitch.org/13007 and the XML response log in 
> http://pastebin.freeswitch.org/13007. Thanks for your help,
> 

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From davis.erwin at gmail.com  Wed May 19 07:39:53 2010
From: davis.erwin at gmail.com (Erwin Davis)
Date: Wed, 19 May 2010 10:39:53 -0400
Subject: [Freeswitch-users] curl_xml
In-Reply-To: <DF566DB4-AA3D-468C-A746-93F57BFED1D3@freeswitch.org>
References: <AANLkTiltmx-qzAAgkQjUcCAGsAMp9_X573zbQic24pf7@mail.gmail.com>
	<DF566DB4-AA3D-468C-A746-93F57BFED1D3@freeswitch.org>
Message-ID: <AANLkTilaaV-d4snb6g1_7X_2k1-PgFDOap1szZ1VlngA@mail.gmail.com>

the xml log file is in
http://pastebin.freeswitch.org/13006<http://pastebin.freeswitch.org/13007>

The webserver sent back the response as below,

   1.   <domain name="192.168.1.31">
   2.                     <user id="1018">
   3.                         <params>
   4.                             <param name="password" value="1018"/>
   5.                         </params>
   6.                     </user>
   7.                 </domain>
   8.



On Wed, May 19, 2010 at 10:25 AM, Brian West <brian at freeswitch.org> wrote:

> What exactly are you returning to freeswitch?  What does the resulting XML
> look like?
>
> /b
>
> On May 19, 2010, at 9:21 AM, Erwin Davis wrote:
>
> Hi, I am trying to use curl_xml to dynamically send back the response for
> directory query from a webserver.
> But when the FS received the response from the web server, it kept sending
> 403 forbidden response to the sip client (x-lite).
> I post the FS log from  the console in
> http://pastebin.freeswitch.org/13007 and the XML response log in
> http://pastebin.freeswitch.org/13007. Thanks for your help,
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From davis.erwin at gmail.com  Wed May 19 07:40:47 2010
From: davis.erwin at gmail.com (Erwin Davis)
Date: Wed, 19 May 2010 10:40:47 -0400
Subject: [Freeswitch-users] curl_xml
In-Reply-To: <DF566DB4-AA3D-468C-A746-93F57BFED1D3@freeswitch.org>
References: <AANLkTiltmx-qzAAgkQjUcCAGsAMp9_X573zbQic24pf7@mail.gmail.com>
	<DF566DB4-AA3D-468C-A746-93F57BFED1D3@freeswitch.org>
Message-ID: <AANLkTimFLCMqXYm27dq9oHPVtJ3pdldblhXcaOGu31BZ@mail.gmail.com>

the xml log file is in
http://pastebin.freeswitch.org/13006<http://pastebin.freeswitch.org/13007>

The webserver sent back the response as below,

   1.   <domain name="192.168.1.31">
   2.                     <user id="1018">
   3.                         <params>
   4.                             <param name="password" value="1018"/>
   5.                         </params>
   6.                     </user>
   7.                 </domain>
   8.



On Wed, May 19, 2010 at 10:25 AM, Brian West <brian at freeswitch.org> wrote:

> What exactly are you returning to freeswitch?  What does the resulting XML
> look like?
>
> /b
>
> On May 19, 2010, at 9:21 AM, Erwin Davis wrote:
>
> Hi, I am trying to use curl_xml to dynamically send back the response for
> directory query from a webserver.
> But when the FS received the response from the web server, it kept sending
> 403 forbidden response to the sip client (x-lite).
> I post the FS log from  the console in
> http://pastebin.freeswitch.org/13007 and the XML response log in
> http://pastebin.freeswitch.org/13007. Thanks for your help,
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Wed May 19 07:42:02 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 19 May 2010 09:42:02 -0500
Subject: [Freeswitch-users] curl_xml
In-Reply-To: <AANLkTimFLCMqXYm27dq9oHPVtJ3pdldblhXcaOGu31BZ@mail.gmail.com>
References: <AANLkTiltmx-qzAAgkQjUcCAGsAMp9_X573zbQic24pf7@mail.gmail.com>
	<DF566DB4-AA3D-468C-A746-93F57BFED1D3@freeswitch.org>
	<AANLkTimFLCMqXYm27dq9oHPVtJ3pdldblhXcaOGu31BZ@mail.gmail.com>
Message-ID: <03E6E212-1AF1-4CB3-8521-FBAD1DA53B7B@freeswitch.org>

Yes that is wrong you're missing some stuff please read: http://wiki.freeswitch.org/wiki/Xml_curl#Response

/b

On May 19, 2010, at 9:40 AM, Erwin Davis wrote:

> the xml log file is in http://pastebin.freeswitch.org/13006
> 
> The webserver sent back the response as below,
>   <domain name="192.168.1.31">
>                     <user id="1018">
>                         <params>
>                             <param name="password" value="1018"/>
>                         </params>
>                     </user>
>                 </domain> 
>  
> 

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From max.clark at gmail.com  Wed May 19 07:55:34 2010
From: max.clark at gmail.com (Max Clark)
Date: Wed, 19 May 2010 07:55:34 -0700
Subject: [Freeswitch-users] Settings for maximum performance
Message-ID: <AANLkTin9UH9IYp4ceb-oSnZ-fBdO8qbNgPzW8MnIjHko@mail.gmail.com>

Hi all,

We are evaluating FreeSWITCH as a replacement for our current
commercial SBCs - before anyone points it out, we need a B2BUA for
sane CDR generation (aka billing) and there is some ideas for future
functionality that would leverage more of the FreeSWITCH platform.

That being said we are currently forcing static IPs, disabling
registration and presence and setting inbound-late-negotiation for the
sip profiles. "Client" gateways are being authenticated using remote
IP ACLs and a dial string prefix. The dialplan looks like this...

http://dpaste.com/196561/

The box in question has two Intel Xeon 5130 dual core processors w/ 4
GB Ram (being upgraded, trying to figure out how much ram we should
have). Disks are 4x 15k RPM SCSI in a Raid 10. Operating system is
CentOS 5.4 x86_64.

FreeSWITCH is being started with "-nc -nonat -waste".

What things should we be looking at to squeeze as much performance as
possible out of these boxes?

Thanks,
Max


[root at fs-sbc02 freeswitch]# vmstat 3
procs -----------memory---------- ---swap-- -----io---- --system--
-----cpu------
 r  b   swpd   free   buff  cache   si   so    bi    bo   in   cs us sy id wa st
 4  0    112  25220 237172 2829060    0    0     0    10    2    2  0
0 99  0  0
 0  1    112  48412 236400 2800596    0    0     0  2408 13298 212312
20 14 63  3  0
25  0    112  46024 236444 2803632    0    0     0  1172 13245 219590
15 13 69  4  0
32  0    112  45784 236512 2808200    0    0     0  2839 13153 164973
21 14 61  3  0
63  1    112  37880 236544 2811808    0    0     0  2884 13048 199455
21 16 60  4  0
 1  0    112  37488 236588 2815612    0    0     0  4000 13201 142378
16 17 50 17  0
 0  0    112  42648 236616 2818376    0    0     0  2125 13171 107533
13 13 70  3  0
 1  0    112  47904 236636 2820704    0    0     0  1989 13400 86644
10 11 72  6  0
 6  0    112  50372 236652 2822200    0    0     0  1585 13339 69289
8  9 78  5  0


From kris at kriskinc.com  Wed May 19 08:45:55 2010
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 19 May 2010 11:45:55 -0400
Subject: [Freeswitch-users] Settings for maximum performance
In-Reply-To: <AANLkTin9UH9IYp4ceb-oSnZ-fBdO8qbNgPzW8MnIjHko@mail.gmail.com>
References: <AANLkTin9UH9IYp4ceb-oSnZ-fBdO8qbNgPzW8MnIjHko@mail.gmail.com>
Message-ID: <AANLkTinN17P0Y1gX_dnaY4t16QuFQYFmbEjjWVdnwwNp@mail.gmail.com>

On Wed, May 19, 2010 at 10:55 AM, Max Clark <max.clark at gmail.com> wrote:
> Hi all,
>
> We are evaluating FreeSWITCH as a replacement for our current
> commercial SBCs - before anyone points it out, we need a B2BUA for
> sane CDR generation (aka billing) and there is some ideas for future
> functionality that would leverage more of the FreeSWITCH platform.
>
> That being said we are currently forcing static IPs, disabling
> registration and presence and setting inbound-late-negotiation for the
> sip profiles. "Client" gateways are being authenticated using remote
> IP ACLs and a dial string prefix. The dialplan looks like this...
>
> http://dpaste.com/196561/
>
> The box in question has two Intel Xeon 5130 dual core processors w/ 4
> GB Ram (being upgraded, trying to figure out how much ram we should
> have). Disks are 4x 15k RPM SCSI in a Raid 10. Operating system is
> CentOS 5.4 x86_64.
>
> FreeSWITCH is being started with "-nc -nonat -waste".
>
> What things should we be looking at to squeeze as much performance as
> possible out of these boxes?
>
> Thanks,
> Max

  Instead of using -waste on the CLI you should be setting the correct
ulimit parameters in your init script:

	ulimit -c unlimited
	ulimit -d unlimited
	ulimit -f unlimited
	ulimit -i unlimited
	ulimit -n 999999
	ulimit -q unlimited
	ulimit -u unlimited
	ulimit -v unlimited
	ulimit -x unlimited
	ulimit -s 240
	ulimit -l unlimited
	ulimit -a

  Something like that, at least.  It sounds like you've already done
most of this, but:

- Disable all NAT/STUN/etc handling in your profiles.  Your FreeSWITCH
instance is not behind NAT (you can still detect NAT from your
endpoints)
- Check, check, double check your SIP profile settings.  Session
timers?  Comfort noise?  TCP?  Do you need these?
- Increase the maximum number of sessions and sessions per second in
switch.conf.  Make sure the RTP port ranges "jive" with what your
network expects.
- Look at the timer test results when FreeSWITCH starts up.  Make sure
its not working too hard just to maintain a timer.
- Disable any unneeded modules in modules.conf.
- Don't use proxy_media.  You probably don't really want it.
- Use as many SIP profiles as you can (I believe each SIP profile runs
in its own thread).
- Make sure APIC, MSI/MSI-X, etc are all working.  Make sure you've
got decent NICs and drivers loaded.
- Email consulting at freeswitch.org :)

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From msc at freeswitch.org  Wed May 19 09:43:51 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 19 May 2010 09:43:51 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call Starting Shortly!
Message-ID: <AANLkTil4Ucvm9SVaswaPPO6WbbaRuNHf_hrwu1omriau@mail.gmail.com>

C'mon down!

http://wiki.freeswitch.org/wiki/FS_weekly_2010_05_19

We have lots of stuff to talk about. :)

-MC
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From ranjtech at gmail.com  Wed May 19 09:50:26 2010
From: ranjtech at gmail.com (RR)
Date: Wed, 19 May 2010 12:50:26 -0400
Subject: [Freeswitch-users] FS on Sparc based Architecture
In-Reply-To: <A30B4BDA-B88D-4B57-B77D-E6E78DD291B9@jerris.com>
References: <AANLkTiklyJ-DwEyRxcEwd2mpvDqrBucZANCR3U2bGuFh@mail.gmail.com>
	<A30B4BDA-B88D-4B57-B77D-E6E78DD291B9@jerris.com>
Message-ID: <AANLkTikCrkFdi0phGQJ-6kYSBI0vDUSulHUGsLJ1XJwN@mail.gmail.com>

Ok. Thanks Mike. Will give it a shot and let you guys know how it works
out.

On Wed, May 19, 2010 at 12:29 AM, Michael Jerris <mike at jerris.com> wrote:

> It should work fine, yes.
>
> On May 18, 2010, at 4:14 PM, RR wrote:
>
> Hello List,
>
> I did a bit of hunting here and there but couldn't find anything that
> mentions any one having any experience/success/failure with
> compiling/building FS under Solaris (other than
> http://wiki.freeswitch.org/wiki/Installation_Guide#Solaris) running over a
> Sparc based architecture. For all I know, it may have nothing to do with it
> being Solaris on Sparc or Intel but does anyone know and can verify for sure
> that using the instructions on the Wiki, FS can be built for Solaris on
> Sparc and will/does stable and seamlessly as it does on Linux etc?
>
> Sorry if this is a silly question but just needed to ask for my own sanity
> :)
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dftoro at yahoo.com  Wed May 19 10:33:45 2010
From: dftoro at yahoo.com (Diego Toro)
Date: Wed, 19 May 2010 10:33:45 -0700 (PDT)
Subject: [Freeswitch-users] TLS on Windows
In-Reply-To: <0B49099E-83DA-4324-827C-E93F52A05FBB@jerris.com>
Message-ID: <798207.36609.qm@web33505.mail.mud.yahoo.com>

Hi MikeJ,

I can work in this subject, I need to know that is required to do.


Diego Toro 
http://lacarretade.blogspot.com/


--- On Wed, 5/19/10, Michael Jerris <mike at jerris.com> wrote:

> From: Michael Jerris <mike at jerris.com>
> Subject: Re: [Freeswitch-users] TLS on Windows
> To: freeswitch-users at lists.freeswitch.org
> Date: Wednesday, May 19, 2010, 8:41 AM
> That patch used static compiled
> openssl.? This means the openssl could be built against
> a different version of the c runtime, which can cause
> issues.? An acceptable patch would have to build
> openssl as well.
> 
> Mike
> 
> On May 19, 2010, at 2:37 AM, Peter Olsson wrote:
> 
> > Someone tried to do a patch once (FSBUILD-70), but I
> don't think it was ever accepted. It would probably need
> some more work.
> > 
> > /Peter
> > 
> > 
> > -----Ursprungligt meddelande-----
> > Fr?n: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org]
> F?r Michael Jerris
> > Skickat: den 19 maj 2010 01:45
> > Till: freeswitch-users at lists.freeswitch.org
> > ?mne: Re: [Freeswitch-users] TLS on Windows
> > 
> > No one ever made the sofia build on windows work with
> openssl so it is not available until we get patches to add
> this.
> > 
> > Mike
> > 
> > On May 18, 2010, at 6:18 PM, Diego Toro wrote:
> > 
> >> I have the certificates, this isn't my
> problem.? Sofia says some like protocol is not
> supported.
> >> 
> >> Thank you
> >> 
> >> Diego Toro
> >> http://lacarretade.blogspot.com/
> >> 
> >> 
> >> --- On Tue, 5/18/10, leo.zibi at gmail.com
> <leo.zibi at gmail.com>
> wrote:
> >> 
> >>> From: leo.zibi at gmail.com
> <leo.zibi at gmail.com>
> >>> Subject: Re: [Freeswitch-users] TLS on
> Windows
> >>> To: freeswitch-users at lists.freeswitch.org
> >>> Date: Tuesday, May 18, 2010, 3:38 PM
> >>> Hi,
> >>> 
> >>> http://wiki.freeswitch.org/wiki/Generating_TLS_certificates_(win32)
> >>> 
> >>> -- 
> >>> Regards
> >>> P.
> >>> 
> >>> Diego Toro wrote:
> >>>> Greetings,
> >>>> 
> >>>> I have been interested about TLS on
> Windows and
> >>> FreeSWITCH, after web search and mailing list,
> I don't know
> >>> if is possible using FreeSWITCH on Windows to
> have security
> >>> communication with TLS. 
> >>>> 
> >>>> I made and installed OpenSSL and I have
> tried build
> >>> FreeSWITCH (libsofia_sip_ua_static) using
> OpenSSL, I created
> >>> the certificates and put it on conf/ssl
> directory, but the
> >>> internal profile faults when it's created, the
> sofia log
> >>> says some like protocol is not supported.
> >>>> 
> >>>> I have lastest git branch 1.0.head,
> windows 7 32
> >>> bits.
> >>>> 
> >>>> I appreciate any suggestion
> > 
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > 
> > !DSPAM:4bf3288332932133842177!
> > 
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


      


From kris at kriskinc.com  Wed May 19 10:58:25 2010
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 19 May 2010 13:58:25 -0400
Subject: [Freeswitch-users] mod_dptools/redirect and 300 Multiple Choices
Message-ID: <AANLkTik7DAQjIA6XvpekW_9aSBPhDozvtWTiR-BGkJb3@mail.gmail.com>

Hello everyone,

  I'd like to use redirect to send 300 Multiple Choices (with multiple
contacts) instead of 302 Moved Temporarily with one contact.  I tried
using redirect and separating multiple contacts with spaces but FS
just sent out a 500 :(.  Is this possible?  If not, how hard would it
be to implement this feature?

Thanks!

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From brian at freeswitch.org  Wed May 19 11:09:32 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 19 May 2010 13:09:32 -0500
Subject: [Freeswitch-users] mod_dptools/redirect and 300 Multiple Choices
In-Reply-To: <AANLkTik7DAQjIA6XvpekW_9aSBPhDozvtWTiR-BGkJb3@mail.gmail.com>
References: <AANLkTik7DAQjIA6XvpekW_9aSBPhDozvtWTiR-BGkJb3@mail.gmail.com>
Message-ID: <ADC2C102-DB14-407B-A87F-04F054F96D9B@freeswitch.org>

The issue is that redirect uses SIP_302_MOVED_TEMPORARILY instead of SIP_300_MULTIPLE_CHOICES so a new option/api would have to be added to allow us to send a 300.

/b

On May 19, 2010, at 12:58 PM, Kristian Kielhofner wrote:

> Hello everyone,
> 
>  I'd like to use redirect to send 300 Multiple Choices (with multiple
> contacts) instead of 302 Moved Temporarily with one contact.  I tried
> using redirect and separating multiple contacts with spaces but FS
> just sent out a 500 :(.  Is this possible?  If not, how hard would it
> be to implement this feature?
> 
> Thanks!
> 
> -- 
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mike at jerris.com  Wed May 19 11:31:35 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 19 May 2010 14:31:35 -0400
Subject: [Freeswitch-users] mod_dptools/redirect and 300 Multiple Choices
In-Reply-To: <ADC2C102-DB14-407B-A87F-04F054F96D9B@freeswitch.org>
References: <AANLkTik7DAQjIA6XvpekW_9aSBPhDozvtWTiR-BGkJb3@mail.gmail.com>
	<ADC2C102-DB14-407B-A87F-04F054F96D9B@freeswitch.org>
Message-ID: <E33F0FBF-C887-42BB-A927-D43FF2F66216@jerris.com>

We could do it maybe with catching the multiple responses in a redirect and automatically doing a 300 instead of 302 or some special handling in respond?

Mike


On May 19, 2010, at 2:09 PM, Brian West wrote:

> The issue is that redirect uses SIP_302_MOVED_TEMPORARILY instead of SIP_300_MULTIPLE_CHOICES so a new option/api would have to be added to allow us to send a 300.
> 
> /b
> 
> On May 19, 2010, at 12:58 PM, Kristian Kielhofner wrote:
> 
>> Hello everyone,
>> 
>> I'd like to use redirect to send 300 Multiple Choices (with multiple
>> contacts) instead of 302 Moved Temporarily with one contact.  I tried
>> using redirect and separating multiple contacts with spaces but FS
>> just sent out a 500 :(.  Is this possible?  If not, how hard would it
>> be to implement this feature?
>> 
>> Thanks!
>> 
>> -- 
>> Kristian Kielhofner
>> http://www.astlinux.org
>> http://blog.krisk.org
>> http://www.star2star.com
>> http://www.submityoursip.com
>> http://www.voalte.com
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From kris at kriskinc.com  Wed May 19 11:39:08 2010
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 19 May 2010 14:39:08 -0400
Subject: [Freeswitch-users] mod_dptools/redirect and 300 Multiple Choices
In-Reply-To: <ADC2C102-DB14-407B-A87F-04F054F96D9B@freeswitch.org>
References: <AANLkTik7DAQjIA6XvpekW_9aSBPhDozvtWTiR-BGkJb3@mail.gmail.com>
	<ADC2C102-DB14-407B-A87F-04F054F96D9B@freeswitch.org>
Message-ID: <AANLkTikTtQfGNO_nKPbj8Ay3nmY7zbLKCOR996BiftzW@mail.gmail.com>

On Wed, May 19, 2010 at 2:09 PM, Brian West <brian at freeswitch.org> wrote:
> The issue is that redirect uses SIP_302_MOVED_TEMPORARILY instead of SIP_300_MULTIPLE_CHOICES so a new option/api would have to be added to allow us to send a 300.
>
> /b

  Hmmm...  I might just try to get away with 302 and specifying
multiple contacts separated with a comma.  Trying this appears to work
but the resulting header is not RFC compliant:

Contact: "unknown" <sip:1 at test.com;q=2;lcr=foo, sip:2 at test.com;q=1;lcr=foo>

should be:

Contact: "unknown" <sip:1 at test.com;q=2;lcr=foo>, <sip:2 at test.com;q=1;lcr=foo>

  Each URI should be enclosed in separate <>.  How hard would it be to
make that change?

  BTW, if you haven't guessed by now I'm experimenting with making
FreeSWITCH a standalone LCR server with mod_lcr.  I'm just curious,
mostly :).

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From kris at kriskinc.com  Wed May 19 11:42:05 2010
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 19 May 2010 14:42:05 -0400
Subject: [Freeswitch-users] mod_dptools/redirect and 300 Multiple Choices
In-Reply-To: <E33F0FBF-C887-42BB-A927-D43FF2F66216@jerris.com>
References: <AANLkTik7DAQjIA6XvpekW_9aSBPhDozvtWTiR-BGkJb3@mail.gmail.com>
	<ADC2C102-DB14-407B-A87F-04F054F96D9B@freeswitch.org>
	<E33F0FBF-C887-42BB-A927-D43FF2F66216@jerris.com>
Message-ID: <AANLkTin3QYwAU_mU9nqiGbVnfL51t0Wvc2iWiCVRjnqk@mail.gmail.com>

On Wed, May 19, 2010 at 2:31 PM, Michael Jerris <mike at jerris.com> wrote:
> We could do it maybe with catching the multiple responses in a redirect and automatically doing a 300 instead of 302 or some special handling in respond?
>
> Mike

300 would probably be better (more similar to other implementations)
but 302 will probably work as many stacks treat them interchangeably.
As long as they can parse multiple Contacts I suppose it doesn't
really matter...

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From brian at freeswitch.org  Wed May 19 11:44:02 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 19 May 2010 13:44:02 -0500
Subject: [Freeswitch-users] mod_dptools/redirect and 300 Multiple Choices
In-Reply-To: <AANLkTikTtQfGNO_nKPbj8Ay3nmY7zbLKCOR996BiftzW@mail.gmail.com>
References: <AANLkTik7DAQjIA6XvpekW_9aSBPhDozvtWTiR-BGkJb3@mail.gmail.com>
	<ADC2C102-DB14-407B-A87F-04F054F96D9B@freeswitch.org>
	<AANLkTikTtQfGNO_nKPbj8Ay3nmY7zbLKCOR996BiftzW@mail.gmail.com>
Message-ID: <DC75CF0A-17C5-48A5-AD9D-8BBC9D78BFD7@freeswitch.org>

Bounty would get it done quickly I suspect.

/b

On May 19, 2010, at 1:39 PM, Kristian Kielhofner wrote:

>  Each URI should be enclosed in separate <>.  How hard would it be to
> make that change?
> 
>  BTW, if you haven't guessed by now I'm experimenting with making
> FreeSWITCH a standalone LCR server with mod_lcr.  I'm just curious,
> mostly :).



From jan.berger at video24.no  Wed May 19 11:50:01 2010
From: jan.berger at video24.no (Jan Berger)
Date: Wed, 19 May 2010 20:50:01 +0200
Subject: [Freeswitch-users] SCTP for Windows
In-Reply-To: <21B967A9-3874-419E-9CFE-827DF2F6EBD9@jerris.com>
References: <527B81D4113C455A964AAE20C28C3D49@dell9400>
	<B80AF699-A323-4031-8900-3AD3081912CB@jerris.com>
	<20100518164235.3F93FD6247B00@bmail02.one.com>
	<9FFF1D7C542C46A2B98D5B8E787E7E4A@dell9400>
	<21B967A9-3874-419E-9CFE-827DF2F6EBD9@jerris.com>
Message-ID: <FBE1BE64D091464883724166B7C1407F@dell9400>

Just to let you know I got this working with a bit help from the Author.

 

Jan

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Jerris
Sent: 19. mai 2010 01:50
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] SCTP for Windows

 

http://sourceforge.net/projects/sctpdrv/ seems to be the one.  It makes me
think maybe its time to come up with combining a few of these offerings to
have a single sctp api with a good license across multiple operating
systems.  Does anyone know any of the key players in any of these open
source stacks?

 

Mike

 

On May 18, 2010, at 4:07 PM, Jan Berger wrote:





You need to download this from sourceforge so you get the english install,
newest version and doc.

 

The source I looked at say this is a Cisco sctp stack ???

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jan
Berger
Sent: 18. mai 2010 18:43
To: freeswitch-users at lists.freeswitch.org;
freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] SCTP for Windows

 

Thanks mate,

 

Japanese ? Might actually work then :)

 

There is a German library as well, but it says GPL. 

 

Jan 

On May 18, 2010 16:42 "Michael Jerris"  <mailto:mike at jerris.com>
<mike at jerris.com> wrote:

http://www.co-conv.jp/en/product/sctpDrv/20081224/ 

 

On May 18, 2010, at 8:09 AM, Jan Berger wrote:






Hi list,

 

Does anyone know about a proper SCTP library for Windows that is reliable
and with a compatible licens?

 

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



 

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From sos at sokhapkin.dyndns.org  Wed May 19 11:58:13 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 19 May 2010 14:58:13 -0400
Subject: [Freeswitch-users] mod_dptools/redirect and 300 Multiple Choices
In-Reply-To: <DC75CF0A-17C5-48A5-AD9D-8BBC9D78BFD7@freeswitch.org>
References: <AANLkTik7DAQjIA6XvpekW_9aSBPhDozvtWTiR-BGkJb3@mail.gmail.com>
	<AANLkTikTtQfGNO_nKPbj8Ay3nmY7zbLKCOR996BiftzW@mail.gmail.com>
	<DC75CF0A-17C5-48A5-AD9D-8BBC9D78BFD7@freeswitch.org>
Message-ID: <201005191458.14068.sos@sokhapkin.dyndns.org>

Review and commit DP-16 patch first please :-)

On Wednesday 19 May 2010, Brian West wrote:
> Bounty would get it done quickly I suspect.
> 
> /b
> 
> On May 19, 2010, at 1:39 PM, Kristian Kielhofner wrote:
> >  Each URI should be enclosed in separate <>.  How hard would it be to
> > make that change?
> >
> >  BTW, if you haven't guessed by now I'm experimenting with making
> > FreeSWITCH a standalone LCR server with mod_lcr.  I'm just curious,
> > mostly :).
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From kris at kriskinc.com  Wed May 19 12:53:02 2010
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 19 May 2010 15:53:02 -0400
Subject: [Freeswitch-users] mod_dptools/redirect and 300 Multiple Choices
In-Reply-To: <DC75CF0A-17C5-48A5-AD9D-8BBC9D78BFD7@freeswitch.org>
References: <AANLkTik7DAQjIA6XvpekW_9aSBPhDozvtWTiR-BGkJb3@mail.gmail.com>
	<ADC2C102-DB14-407B-A87F-04F054F96D9B@freeswitch.org>
	<AANLkTikTtQfGNO_nKPbj8Ay3nmY7zbLKCOR996BiftzW@mail.gmail.com>
	<DC75CF0A-17C5-48A5-AD9D-8BBC9D78BFD7@freeswitch.org>
Message-ID: <AANLkTilLdN-2DkmPBkccMB8B60V-X2Cq0olPDssiyDRx@mail.gmail.com>

http://jira.freeswitch.org/browse/BOUNTY-18

On Wed, May 19, 2010 at 2:44 PM, Brian West <brian at freeswitch.org> wrote:
> Bounty would get it done quickly I suspect.
>
> /b
>

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From max.clark at gmail.com  Wed May 19 13:06:13 2010
From: max.clark at gmail.com (Max Clark)
Date: Wed, 19 May 2010 13:06:13 -0700
Subject: [Freeswitch-users] Settings for maximum performance
In-Reply-To: <AANLkTinN17P0Y1gX_dnaY4t16QuFQYFmbEjjWVdnwwNp@mail.gmail.com>
References: <AANLkTin9UH9IYp4ceb-oSnZ-fBdO8qbNgPzW8MnIjHko@mail.gmail.com>
	<AANLkTinN17P0Y1gX_dnaY4t16QuFQYFmbEjjWVdnwwNp@mail.gmail.com>
Message-ID: <AANLkTil77yRj5S3U0pgiqfqouvlfibt_3G3Ou1MLPlnu@mail.gmail.com>

On Wed, May 19, 2010 at 8:45 AM, Kristian Kielhofner <kris at kriskinc.com> wrote:
> ? ? ? ?ulimit -c unlimited
> ? ? ? ?ulimit -d unlimited
> ? ? ? ?ulimit -f unlimited
> ? ? ? ?ulimit -i unlimited
> ? ? ? ?ulimit -n 999999
> ? ? ? ?ulimit -q unlimited
> ? ? ? ?ulimit -u unlimited
> ? ? ? ?ulimit -v unlimited
> ? ? ? ?ulimit -x unlimited
> ? ? ? ?ulimit -s 240
> ? ? ? ?ulimit -l unlimited
> ? ? ? ?ulimit -a

Okay, set and removed -waste from init script.

> - Disable all NAT/STUN/etc handling in your profiles. ?Your FreeSWITCH
> instance is not behind NAT (you can still detect NAT from your
> endpoints)

So my sip profile now looks like this:

    <param name="rtp-ip" value="$${local_ip_v4}"/>
    <param name="sip-ip" value="$${local_ip_v4}"/>
    <param name="ext-rtp-ip" value="$${local_ip_v4}"/>
    <param name="ext-sip-ip" value="$${local_ip_v4}"/>
    <param name="stun-enabled" value="false"/>
    <param name="stun-auto-disable" value="true"/>

I didn't find anything in the sofia configuration for disabling nat
which is why I have the -nonat on startup now. Is there another option
I should be setting?

> - Check, check, double check your SIP profile settings. ?Session
> timers? ?Comfort noise? ?TCP? ?Do you need these?

Looking into these settings - what's sane for my usage?

> - Increase the maximum number of sessions and sessions per second in
> switch.conf. ?Make sure the RTP port ranges "jive" with what your
> network expects.

How do I size sessions & sessions per second based on the machine cpu/ram?

> - Look at the timer test results when FreeSWITCH starts up. ?Make sure
> its not working too hard just to maintain a timer.

Looking through the freeswitch.log file I do not see any output on
timers - should I be looking somewhere else?

> - Disable any unneeded modules in modules.conf.

Check

> - Don't use proxy_media. ?You probably don't really want it.

I have proxy_media because we want to keep the RTP streams through our
boxes (topology hiding), we were looking for better t.38 handling, we
don't want to transcode, and it seemed okay to let the endpoints
negotiate g.711 or g.729 on their own.

> - Use as many SIP profiles as you can (I believe each SIP profile runs
> in its own thread).

This one is tough, we'll take another look at it.

> - Make sure APIC, MSI/MSI-X, etc are all working. ?Make sure you've
> got decent NICs and drivers loaded.

I'm assuming that Dell PowerEdge servers aren't going to be an issue.

> - Email consulting at freeswitch.org :)

Trying to knock out all the "easy" stuff first.

Thank you very much for the pointers.

-Max


From david.ponzone at gmail.com  Wed May 19 13:20:16 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Wed, 19 May 2010 22:20:16 +0200
Subject: [Freeswitch-users] Settings for maximum performance
In-Reply-To: <AANLkTil77yRj5S3U0pgiqfqouvlfibt_3G3Ou1MLPlnu@mail.gmail.com>
References: <AANLkTin9UH9IYp4ceb-oSnZ-fBdO8qbNgPzW8MnIjHko@mail.gmail.com>
	<AANLkTinN17P0Y1gX_dnaY4t16QuFQYFmbEjjWVdnwwNp@mail.gmail.com>
	<AANLkTil77yRj5S3U0pgiqfqouvlfibt_3G3Ou1MLPlnu@mail.gmail.com>
Message-ID: <F7CBE1B8-C808-42E7-A369-E3E244386D31@gmail.com>

>
>
>> - Check, check, double check your SIP profile settings.  Session
>> timers?  Comfort noise?  TCP?  Do you need these?
>
> Looking into these settings - what's sane for my usage?

Session timers are useful if you have issues with ghost calls.
If you think you're "ghost-free", you may disable it.

>> - Increase the maximum number of sessions and sessions per second in
>> switch.conf.  Make sure the RTP port ranges "jive" with what your
>> network expects.
>
> How do I size sessions & sessions per second based on the machine  
> cpu/ram?

That's a good question.
I think you'll probably need to discover that yourself from experience.
For a box like the one you described, I would say 2000 calls and 100  
cps is a good start.

>> - Look at the timer test results when FreeSWITCH starts up.  Make  
>> sure
>> its not working too hard just to maintain a timer.
>
> Looking through the freeswitch.log file I do not see any output on
> timers - should I be looking somewhere else?

This shows up during startup.


David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.

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From kris at kriskinc.com  Wed May 19 13:22:57 2010
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 19 May 2010 16:22:57 -0400
Subject: [Freeswitch-users] Settings for maximum performance
In-Reply-To: <AANLkTil77yRj5S3U0pgiqfqouvlfibt_3G3Ou1MLPlnu@mail.gmail.com>
References: <AANLkTin9UH9IYp4ceb-oSnZ-fBdO8qbNgPzW8MnIjHko@mail.gmail.com>
	<AANLkTinN17P0Y1gX_dnaY4t16QuFQYFmbEjjWVdnwwNp@mail.gmail.com>
	<AANLkTil77yRj5S3U0pgiqfqouvlfibt_3G3Ou1MLPlnu@mail.gmail.com>
Message-ID: <AANLkTimWzXGLaXyeJJZWdl56CJVbUMgev7Sps-mblSbi@mail.gmail.com>

On Wed, May 19, 2010 at 4:06 PM, Max Clark <max.clark at gmail.com> wrote:
>
> So my sip profile now looks like this:
>
> ? ?<param name="rtp-ip" value="$${local_ip_v4}"/>
> ? ?<param name="sip-ip" value="$${local_ip_v4}"/>
> ? ?<param name="ext-rtp-ip" value="$${local_ip_v4}"/>
> ? ?<param name="ext-sip-ip" value="$${local_ip_v4}"/>
> ? ?<param name="stun-enabled" value="false"/>
> ? ?<param name="stun-auto-disable" value="true"/>
>
> I didn't find anything in the sofia configuration for disabling nat
> which is why I have the -nonat on startup now. Is there another option
> I should be setting?

  Not really.  Remove the ext-*-ip params and you should be fine.

> Looking into these settings - what's sane for my usage?

  You probably don't need any of them.  Disable timers, CNG, and TCP.

> How do I size sessions & sessions per second based on the machine cpu/ram?

  You'll have to find out from testing with your config but I'm
assuming you'll want to go beyond the defaults.  You could probably
start by doubling or tripling the default values.

> Looking through the freeswitch.log file I do not see any output on
> timers - should I be looking somewhere else?

  Start FreeSWITCH with the console (-c).  Look at the output in the
first ~20 lines.  You'll see things about timers and tests.  Make sure
it doesn't issue a warning about audio quality and that the offset
isn't too large.

> I have proxy_media because we want to keep the RTP streams through our
> boxes (topology hiding), we were looking for better t.38 handling, we
> don't want to transcode, and it seemed okay to let the endpoints
> negotiate g.711 or g.729 on their own.

  T.38 requires proxy_media but other than that you may be better off
with the default bridging behavior.  It will proxy media and hide your
topology.  You can disable transcoding in the profile and/or have all
kinds of control over codec parameters in the dialplan in combination
with late-negotiation.

> I'm assuming that Dell PowerEdge servers aren't going to be an issue.

  If they're relatively new they should be ok but check
/proc/interrupts to make sure (look for MSI/MSI-X/APIC next to eth0,
eth1, etc).

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From msc at freeswitch.org  Wed May 19 13:44:44 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 19 May 2010 13:44:44 -0700
Subject: [Freeswitch-users] FreeSWITCH Training!
Message-ID: <AANLkTilMLUDBXTRneHImXZOP21PQxd5AvozBJ1nfLnQW@mail.gmail.com>

We are please to let everyone know that official FreeSWITCH training is now
available! More information available here:

http://www.freeswitch.org/node/259

Thanks to Darren Schreiber for organizing the first formal FreeSWITCH
training course.

-Michael
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From kris at kriskinc.com  Wed May 19 15:33:48 2010
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 19 May 2010 18:33:48 -0400
Subject: [Freeswitch-users] mod_dptools/redirect and 300 Multiple Choices
In-Reply-To: <AANLkTilLdN-2DkmPBkccMB8B60V-X2Cq0olPDssiyDRx@mail.gmail.com>
References: <AANLkTik7DAQjIA6XvpekW_9aSBPhDozvtWTiR-BGkJb3@mail.gmail.com>
	<ADC2C102-DB14-407B-A87F-04F054F96D9B@freeswitch.org>
	<AANLkTikTtQfGNO_nKPbj8Ay3nmY7zbLKCOR996BiftzW@mail.gmail.com>
	<DC75CF0A-17C5-48A5-AD9D-8BBC9D78BFD7@freeswitch.org>
	<AANLkTilLdN-2DkmPBkccMB8B60V-X2Cq0olPDssiyDRx@mail.gmail.com>
Message-ID: <AANLkTimGjc21-LJssGi50gBC8v1ON5a2UKrKkuBpWuwF@mail.gmail.com>

Tony got this in, awesome!

One question - the RFC says a UA MAY use multiple Contact headers or
multiple addresses in a Contact field.

As implemented now FreeSWITCH uses multiple Contact headers.  While I
don't see a problem with this I wonder how difficult it would be to
change it to a single Contact header if someone requested.

On Wed, May 19, 2010 at 3:53 PM, Kristian Kielhofner <kris at kriskinc.com> wrote:
> http://jira.freeswitch.org/browse/BOUNTY-18
>

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From brian at freeswitch.org  Wed May 19 15:38:47 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 19 May 2010 17:38:47 -0500
Subject: [Freeswitch-users] mod_dptools/redirect and 300 Multiple Choices
In-Reply-To: <AANLkTimGjc21-LJssGi50gBC8v1ON5a2UKrKkuBpWuwF@mail.gmail.com>
References: <AANLkTik7DAQjIA6XvpekW_9aSBPhDozvtWTiR-BGkJb3@mail.gmail.com>
	<ADC2C102-DB14-407B-A87F-04F054F96D9B@freeswitch.org>
	<AANLkTikTtQfGNO_nKPbj8Ay3nmY7zbLKCOR996BiftzW@mail.gmail.com>
	<DC75CF0A-17C5-48A5-AD9D-8BBC9D78BFD7@freeswitch.org>
	<AANLkTilLdN-2DkmPBkccMB8B60V-X2Cq0olPDssiyDRx@mail.gmail.com>
	<AANLkTimGjc21-LJssGi50gBC8v1ON5a2UKrKkuBpWuwF@mail.gmail.com>
Message-ID: <773BA751-D20F-4A04-A8FA-6598656FCF13@freeswitch.org>

We actually found that if you enable compressed headers it will bunch them into one contact header ;)

/b

On May 19, 2010, at 5:33 PM, Kristian Kielhofner wrote:

> Tony got this in, awesome!
> 
> One question - the RFC says a UA MAY use multiple Contact headers or
> multiple addresses in a Contact field.
> 
> As implemented now FreeSWITCH uses multiple Contact headers.  While I
> don't see a problem with this I wonder how difficult it would be to
> change it to a single Contact header if someone requested.



From kris at kriskinc.com  Wed May 19 15:49:04 2010
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 19 May 2010 18:49:04 -0400
Subject: [Freeswitch-users] mod_dptools/redirect and 300 Multiple Choices
In-Reply-To: <773BA751-D20F-4A04-A8FA-6598656FCF13@freeswitch.org>
References: <AANLkTik7DAQjIA6XvpekW_9aSBPhDozvtWTiR-BGkJb3@mail.gmail.com>
	<ADC2C102-DB14-407B-A87F-04F054F96D9B@freeswitch.org>
	<AANLkTikTtQfGNO_nKPbj8Ay3nmY7zbLKCOR996BiftzW@mail.gmail.com>
	<DC75CF0A-17C5-48A5-AD9D-8BBC9D78BFD7@freeswitch.org>
	<AANLkTilLdN-2DkmPBkccMB8B60V-X2Cq0olPDssiyDRx@mail.gmail.com>
	<AANLkTimGjc21-LJssGi50gBC8v1ON5a2UKrKkuBpWuwF@mail.gmail.com>
	<773BA751-D20F-4A04-A8FA-6598656FCF13@freeswitch.org>
Message-ID: <AANLkTinvFSad8k0GQcCIQw9XQ435KqIiKUNu1uBc0Dp4@mail.gmail.com>

Is that compressed headers or compact headers (or are they the same thing)?

Is this inherent Sofia behavior?  If so, I trust them ;).

On Wed, May 19, 2010 at 6:38 PM, Brian West <brian at freeswitch.org> wrote:
> We actually found that if you enable compressed headers it will bunch them into one contact header ;)
>
> /b

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From brian at freeswitch.org  Wed May 19 15:53:13 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 19 May 2010 17:53:13 -0500
Subject: [Freeswitch-users] mod_dptools/redirect and 300 Multiple Choices
In-Reply-To: <AANLkTinvFSad8k0GQcCIQw9XQ435KqIiKUNu1uBc0Dp4@mail.gmail.com>
References: <AANLkTik7DAQjIA6XvpekW_9aSBPhDozvtWTiR-BGkJb3@mail.gmail.com>
	<ADC2C102-DB14-407B-A87F-04F054F96D9B@freeswitch.org>
	<AANLkTikTtQfGNO_nKPbj8Ay3nmY7zbLKCOR996BiftzW@mail.gmail.com>
	<DC75CF0A-17C5-48A5-AD9D-8BBC9D78BFD7@freeswitch.org>
	<AANLkTilLdN-2DkmPBkccMB8B60V-X2Cq0olPDssiyDRx@mail.gmail.com>
	<AANLkTimGjc21-LJssGi50gBC8v1ON5a2UKrKkuBpWuwF@mail.gmail.com>
	<773BA751-D20F-4A04-A8FA-6598656FCF13@freeswitch.org>
	<AANLkTinvFSad8k0GQcCIQw9XQ435KqIiKUNu1uBc0Dp4@mail.gmail.com>
Message-ID: <52CD42D7-529A-46D1-AADE-A1E58CB1F881@freeswitch.org>

Its compact.. compressed... oh well long day.

/b

On May 19, 2010, at 5:49 PM, Kristian Kielhofner wrote:

> Is that compressed headers or compact headers (or are they the same thing)?
> 
> Is this inherent Sofia behavior?  If so, I trust them ;).



From neilp at cs.stanford.edu  Wed May 19 18:16:06 2010
From: neilp at cs.stanford.edu (Neil Patel)
Date: Wed, 19 May 2010 18:16:06 -0700
Subject: [Freeswitch-users] how do I detect a call has rung once?
In-Reply-To: <AANLkTilt2WwQJmSoOqraooUdREErvKWL_sxi6ZlKiPUY@mail.gmail.com>
References: <u2je5503bdb1005101403u405a5eebo9a2b8c698e9cffaf@mail.gmail.com> 
	<AANLkTilVHAdZDgF3a5Mg2HgXrXOg6hkrLKGb9xUAAdik@mail.gmail.com> 
	<AANLkTikqOpUTWkv9izTLM8CKljgexflNPPEVUxiYzw01@mail.gmail.com> 
	<E08F3AA9-613E-446A-BD22-5B31F38EB78C@freeswitch.org>
	<AANLkTimHH6zzA_-Dal1YkHlSL4gkbp7m8TyoFrukBmB3@mail.gmail.com> 
	<AANLkTinHQnCTPTDtxXRl80Pd9aGMs8V2Xws1z5KeMegU@mail.gmail.com> 
	<AANLkTilcSOp1nfZxSqtndA_HBvKNl1iPrOwJk4uC3P4z@mail.gmail.com> 
	<AANLkTiklAqw6Er_0fCq9CQnt6RyEPY9P0Usg5iAjl4nw@mail.gmail.com> 
	<AANLkTinnTRKaOUI2OHSoi9Lu1jjbSfTxZjFBn_9EIMa1@mail.gmail.com> 
	<AANLkTilt2WwQJmSoOqraooUdREErvKWL_sxi6ZlKiPUY@mail.gmail.com>
Message-ID: <AANLkTilR0VByQMmANUo9Uz4cyH1r2RVcU3xMZP-Kkyxh@mail.gmail.com>

I played around with a couple options and found that issuing a hupall
normal_clearing on the initiated call after a few seconds was the most
reliable way to do this. The next best option I found was hanging up after a
few seconds of receiving CS_EXECUTE state update, though this seemed to
cause the timing to vary.

Are there other events that come over ESL on a PRI call that are good to
anchor to? For some of the suggestions above like "progressing", it isn't
clear what event that corresponds to.

On Thu, May 13, 2010 at 5:14 PM, Neil Patel <neilp at cs.stanford.edu> wrote:

> Thanks Michael, I was looking for code examples to receive events over ESL,
> and found them in libs/esl/.
>
> I'm still not clear on the following:
>
>    1. I have a controller script that originates calls; it runs "show
>    channels" between calls and parses out the number of active channels. If
>    below threshold, originate next call. If not, wait a couple seconds and run
>    "show channels" again, repeating till below threshold. Is this the best way?
>    Can this be done more efficiently through events? How?
>    2. For placing missed calls to SIP endpoints, it sounds like* *I can
>    wait for a signal 180 or 183 to know when to hang up. How do I check for
>    those signals? Is this another job for events?
>    3. For placing missed calls over openzap/PRI, it sounds like I can
>    check for call states PROGRESS or ALERTING? Where are those states from, and
>    how do I check them? Again, can I use events? The other option you gave is
>    checking for endpoint_disposition. This is in the cdr, but how do I use it
>    for detecting when I should hang up for a missed call? Is there an event
>    associated with it? What are the call states it goes through?
>
> -Neil
>
> On Wed, May 12, 2010 at 2:47 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>> On Wed, May 12, 2010 at 11:59 AM, Neil Patel <neilp at cs.stanford.edu>wrote:
>>
>>> Thanks for the tips.
>>>
>>> Where are there examples of using ESL to register and handle events? For
>>> e.g. I don't see any example of how to register for a change in channel
>>> variable endpoint_disposition. Or even to check the CHANNEL_STATE event.
>>>
>>> -Neil
>>>
>>> Neil,
>>
>> Check out the event socket documentation, specifically the "events plain
>> all" and "filter" commands. You can listen to all events or just specific
>> ones:
>> http://wiki.freeswitch.org/wiki/Event_socket
>>
>> If you want to experiment with the event socket and the various events
>> then use fs_cli. Connect with fs_cli and then issue "/log 0" which turns off
>> all debug messages. Now you have basically a raw event socket. Try some of
>> these commands:
>>
>> /event plain all
>>
>> watch the fun messages fly. :)
>>
>> You can filter them as well:
>>
>> /filter Event-Name CHANNEL_STATE
>> /filter Event-Name CHANNEL_HANGUP
>>
>> when you apply a filter you will receive only those things you choose. A
>> it's a "filter in" not filter out. Another way of saying it is "show me
>> events named CHANNEL_STATE" and "show me events named CHANNEL_HANGUP"
>>
>> just note that in your script you will be using the ESL abstractions for
>> these:
>>
>> http://wiki.freeswitch.org/wiki/Esl#ESLconnection_Object
>>
>> I hope that helps! Have fun.
>> -MC
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From cliff at develix.com  Wed May 19 19:02:01 2010
From: cliff at develix.com (Cliff Wells)
Date: Wed, 19 May 2010 19:02:01 -0700
Subject: [Freeswitch-users] Adding variable to be passed via mod_xml_curl
Message-ID: <1274320921.21550.1501.camel@portable-evil>

I'm unable to figure out how to add a custom value to be passed in the
HTTP post.

2010-05-19 21:29:02.028280 [WARNING] ozmod_sangoma_boost.c:1632 RX EVENT: CALL_START:(80) [w1g20] CSid=0 Seq=2995 Cn=[N/A] Cd=[4067150999] Ci=[7025337717] Rdnis=[SMG003-CPC-010-GEN-000-4062249157-003-0-0-1]



-- 



From cliff at develix.com  Wed May 19 19:29:41 2010
From: cliff at develix.com (Cliff Wells)
Date: Wed, 19 May 2010 19:29:41 -0700
Subject: [Freeswitch-users] Adding variable to be passed via mod_xml_curl
Message-ID: <1274322581.21550.1558.camel@portable-evil>

I'm trying to use mod_xml_curl to fetch a dialplan based on the rdnis
and it's turning out to be harder than it should.

I'm unable to figure out how to add a custom value to be passed in the
HTTP post:

2010-05-19 21:29:02.028280 [WARNING] ozmod_sangoma_boost.c:1632 RX EVENT: CALL_START:(80) [w1g20] 
   CSid=0 Seq=2995 Cn=[N/A] Cd=[4067150999] Ci=[7025337717] Rdnis=[SMG003-CPC-010-GEN-000-4062249157-003-0-0-1]


I need to acquire and pass the 4062249157 value from the Rdnis to my
HTTP server script (that's the actual Rdnis, the value in Cd is the LRN
and useless to me).  Unfortunately, for whatever reason, mod_xml_curl
only passes the LRN to my script.

I'm having a difficult time as I can't determine exactly where I can put
anything that will happen before mod_xml_curl is called for the dialplan
(so I can't even use application info, for example, to see the list of
actual variables).   Any advice on how to do this one thing would
probably get me 90% of the way there.

Thanks in advance,
Cliff
-- 




From bwibowo at gmail.com  Wed May 19 19:32:16 2010
From: bwibowo at gmail.com (Budi wibowo)
Date: Thu, 20 May 2010 02:32:16 +0000
Subject: [Freeswitch-users] Billing app
Message-ID: <1492625819-1274322738-cardhu_decombobulator_blackberry.rim.net-320411259-@bda057.bisx.prodap.on.blackberry>

Hi guys 
Any suggestion on good billing platform for FS, minimum for postpaid, but prepaid also ok.

Thx
Budi wibowo 


From max.clark at gmail.com  Wed May 19 19:40:33 2010
From: max.clark at gmail.com (Max Clark)
Date: Wed, 19 May 2010 19:40:33 -0700
Subject: [Freeswitch-users] Settings for maximum performance
In-Reply-To: <AANLkTimWzXGLaXyeJJZWdl56CJVbUMgev7Sps-mblSbi@mail.gmail.com>
References: <AANLkTin9UH9IYp4ceb-oSnZ-fBdO8qbNgPzW8MnIjHko@mail.gmail.com>
	<AANLkTinN17P0Y1gX_dnaY4t16QuFQYFmbEjjWVdnwwNp@mail.gmail.com>
	<AANLkTil77yRj5S3U0pgiqfqouvlfibt_3G3Ou1MLPlnu@mail.gmail.com>
	<AANLkTimWzXGLaXyeJJZWdl56CJVbUMgev7Sps-mblSbi@mail.gmail.com>
Message-ID: <AANLkTinWSgyrIo5aebu_hqFxzeIBRuwOesECyN1pKfRb@mail.gmail.com>

> ?If they're relatively new they should be ok but check
> /proc/interrupts to make sure (look for MSI/MSI-X/APIC next to eth0,
> eth1, etc).

You said earlier to make sure that MSI was working properly - how do I
confirm this? Or is this something I should disable? What is best for
FreeSWITCH?

Thanks,
Max

[mclark at fs-sbc02 ~]$ cat /proc/interrupts
           CPU0       CPU1       CPU2       CPU3
  0: 3969295082          0          0          0    IO-APIC-edge  timer
  1:          3          0          0          0    IO-APIC-edge  i8042
  8:          1          0          0          0    IO-APIC-edge  rtc
  9:          0          0          0          0   IO-APIC-level  acpi
 12:          4          0          0          0    IO-APIC-edge  i8042
 14:         25          0          0          0    IO-APIC-edge  ide0
 66:         57          0          0          0   IO-APIC-level
ehci_hcd:usb1, uhci_hcd:usb2, uhci_hcd:usb4
 74:          0          0          0          0   IO-APIC-level  uhci_hcd:usb3
 82:   11263182    4932428      71397      54669   IO-APIC-level  megasas
 98: 1738518148          0          0          0         PCI-MSI  eth0
NMI:      91918      46871      42712      41086
LOC: 3969292329 3969292268 3969292197 3969292121
ERR:          0
MIS:          0


From cliff at develix.com  Wed May 19 19:52:38 2010
From: cliff at develix.com (Cliff Wells)
Date: Wed, 19 May 2010 19:52:38 -0700
Subject: [Freeswitch-users] Adding variable to be passed via mod_xml_curl
In-Reply-To: <1274322581.21550.1558.camel@portable-evil>
References: <1274322581.21550.1558.camel@portable-evil>
Message-ID: <1274323958.21550.1603.camel@portable-evil>

On Wed, 2010-05-19 at 19:29 -0700, Cliff Wells wrote:

> I'm having a difficult time as I can't determine exactly where I can put
> anything that will happen before mod_xml_curl is called for the dialplan
> (so I can't even use application info, for example, to see the list of
> actual variables).   Any advice on how to do this one thing would
> probably get me 90% of the way there.

Actually, that might not help anyway.

It was suggested to me on IRC to give mod_xml_curl an invalid URL to
cause a variable dump (inbound caller ID obfuscated):

Channel-State: [CS_EXECUTE]
Channel-State-Number: [4]
Channel-Name: [OpenZAP/1:22/4067150999]
Unique-ID: [5ab07bb6-63ba-11df-bd7c-555cb1b118d6]
Call-Direction: [inbound]
Presence-Call-Direction: [inbound]
Answer-State: [ringing]
Channel-Read-Codec-Name: [PCMU]
Channel-Read-Codec-Rate: [8000]
Channel-Write-Codec-Name: [PCMU]
Channel-Write-Codec-Rate: [8000]
Caller-Username: [OpenZAP]
Caller-Dialplan: [XML]
Caller-Caller-ID-Name: [708xxxxxx]
Caller-Caller-ID-Number: [708xxxxxx]
Caller-ANI: [708xxxxxxx]
Caller-Destination-Number: [4067150999]
Caller-Unique-ID: [5ab07bb6-63ba-11df-bd7c-555cb1b118d6]
Caller-Source: [mod_openzap]
Caller-Context: [default]
Caller-Channel-Name: [OpenZAP/1:22/4067150999]
Caller-Profile-Index: [1]
Caller-Profile-Created-Time: [1274323788097303]
Caller-Channel-Created-Time: [1274323788097303]
Caller-Channel-Answered-Time: [0]
Caller-Channel-Progress-Time: [0]
Caller-Channel-Progress-Media-Time: [0]
Caller-Channel-Hangup-Time: [0]
Caller-Channel-Transfer-Time: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
variable_direction: [inbound]
variable_uuid: [5ab07bb6-63ba-11df-bd7c-555cb1b118d6]
variable_read_codec: [PCMU]
variable_read_rate: [8000]
variable_write_codec: [PCMU]
variable_write_rate: [8000]
variable_channel_name: [OpenZAP/1:22/4067150999]
variable_openzap_span_name: [boostss7]
variable_openzap_span_number: [1]
variable_openzap_chan_number: [22]
variable_current_application: [info]

As can be seen, only the LRN (4067150999) is available, so even if I
were able to parse something and set a variable, there doesn't appear to
be anything to parse from.

I guess I'll take it up with the Sangoma folks and see if they can offer
a solution (or point the finger elsewhere).

Regards,
Cliff


-- 




From brian at freeswitch.org  Wed May 19 19:56:26 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 19 May 2010 21:56:26 -0500
Subject: [Freeswitch-users] Adding variable to be passed via mod_xml_curl
In-Reply-To: <1274323958.21550.1603.camel@portable-evil>
References: <1274322581.21550.1558.camel@portable-evil>
	<1274323958.21550.1603.camel@portable-evil>
Message-ID: <2E14A68A-0B92-4BBD-9356-6CA6376D624D@freeswitch.org>

is this not what you want?

Sent from my iPad

On May 19, 2010, at 9:52 PM, Cliff Wells <cliff at develix.com> wrote:

> Caller-Destination-Number: [4067150999]
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From brian at freeswitch.org  Wed May 19 19:57:51 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 19 May 2010 21:57:51 -0500
Subject: [Freeswitch-users] Adding variable to be passed via mod_xml_curl
In-Reply-To: <1274322581.21550.1558.camel@portable-evil>
References: <1274322581.21550.1558.camel@portable-evil>
Message-ID: <F7E4DEF9-0080-4655-BA57-9575834E64AC@freeswitch.org>

I see you'll need to ask Sangoma on this one.

Sent from my iPad

On May 19, 2010, at 9:29 PM, Cliff Wells <cliff at develix.com> wrote:

> 
> I need to acquire and pass the 4062249157 value from the Rdnis to my
> HTTP server script (that's the actual Rdnis, the value in Cd is the LRN
> and useless to me).  Unfortunately, for whatever reason, mod_xml_curl
> only passes the LRN to my script.
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From babak.freeswitch at gmail.com  Wed May 19 22:11:51 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Thu, 20 May 2010 05:11:51 +0000 (UTC)
Subject: [Freeswitch-users] =?utf-8?q?mod=5Fmanaged_bind_problem?=
References: <AANLkTilO75HysCaRkSu6oOmorPsItLhojiEjjRqpqdPW@mail.gmail.com>
Message-ID: <loom.20100520T070859-619@post.gmane.org>

Hi again
After debuggning freeswitch i found out the problem is that nod_index is not 
initialized and
if (node_index <= SWITCH_EVENT_ALL && 
		switch_event_bind_removable(__FILE__, event_id, subclass_name, 
event_handler, this, &enodes[node_index]) == SWITCH_STATUS_SUCCESS) {
		switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "bound to 
%s %s\n", event_name, switch_str_nil(subclass_name));
	
the first part of if condition is always evaluated to false node_index <= 
SWITCH_EVENT_ALL 
when I initialize nod_index to 0 in event consumer constructor it seems working



From shroukkhan at softverk.is  Thu May 20 03:53:10 2010
From: shroukkhan at softverk.is (Shrouk Khan)
Date: Thu, 20 May 2010 17:53:10 +0700
Subject: [Freeswitch-users] nibblebill never hangs up
Message-ID: <AANLkTik1VCDqTy4sFNisPO2HWyKcqBoDNYHdV43FXV2M@mail.gmail.com>

hi ,
i have successfully installed the nibble bill and trying to test billing
depending on the destination , but even though the Cash goes to negetive the
call never hangs up !

here are my conf :

nibblebill.conf
------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------


<configuration name="nibblebill.conf" description="Nibble Billing">
  <settings>
    <!-- See http://wiki.freeswitch.org/wiki/Mod_nibblebill for help with
these options -->

    <!-- Information for connecting to your database -->
    <param name="db_username" value="root"/>
    <param name="db_password" value="fusionpbx"/>
    <param name="db_dsn" value="freeswitch_odbc"/>

    <!-- The database table where your CASH column is located -->
    <param name="db_table" value="v_extensions"/>

    <!-- The column name where we store the value of the account -->
    <param name="db_column_cash" value="Cash"/>

    <!-- The column name for the unique ID identifying the account -->
    <param name="db_column_account" value="extension"/>

    <!-- Custom SQL for loading current balance - overrides column names
         channel vars are interpreted.
         field nibble_balance is used for balance info

    <param name="custom_sql_lookup" value="SELECT cash AS nibble_balance
FROM accounts WHERE account_code='${nibble_account}'"/>
    -->

    <!-- Custom SQL for loading current balance - overrides column names
         channel vars are interpreted.
         nibble_increment is the amount to update

    <param name="custom_sql_save" value="UPDATE accounts SET
cash=cash-${nibble_increment} WHERE account_code='${nibble_account}'"/>
    -->


    <!-- Default heartbeat interval. Set to 'off' for no heartbeat (i.e.
bill only at end of call) -->
    <param name="global_heartbeat" value="30"/>

    <!-- By default, warn a caller when their balance is at $5.00. You can
set this to a negative number. -->
    <param name="lowbal_amt" value="5"/>
    <param name="lowbal_action" value="play ding"/>

    <!-- By default, terminate a caller when their balance hits $0.00. You
can set this to a negative number. -->
    <param name="nobal_amt" value="0.00"/>
    <param name="nobal_action" value="hangup"/>

    <!-- If a call goes beyond a certain dollar amount, flag or terminate it
-->
    <param name="percall_max_amt" value="100"/>
    <param name="percall_action" value="hangup"/>

  </settings>
</configuration>
------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------

an extension called 5400

<include>
  <user id="5400">
    <params>
      <param name="password" value="5400"/>
      <param name="vm-password" value="8975"/>
    </params>
    <variables>
      <variable name="toll_allow" value="domestic,international,local"/>
      <variable name="accountcode" value="5400"/>
      <variable name="user_context" value="default"/>
      <variable name="nibble_account" value="5400"/>
    </variables>
  </user>
</include>

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


finally in the dial plan :

<extension name="ToIcelandPBX.58d2" >
  <condition field="destination_number" expression="^(58.*)">
    <action application="set" data="$${nibble_account}"/>
  </condition>
  <condition field="destination_number" expression="^(5807)$">
    <action application="set" data="nibble_rate=30"/>
    <action application="bridge" data="sofia/gateway/ToIcelandPBX/$1"/>
  </condition>

  <condition field="destination_number" expression="^(5812)$">
    <action application="set" data="nibble_rate=15"/>
    <action application="bridge" data="sofia/gateway/ToIcelandPBX/$1"/>
  </condition>

</extension>


right now the Cash balance for the extension is -192.00 and the extension
can still make calls jsut fine.
Am I missing something here ?

-- 
Regards

Shrouk Khan (Khan)
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From freeswitch-list at puzzled.xs4all.nl  Thu May 20 05:17:28 2010
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Thu, 20 May 2010 14:17:28 +0200
Subject: [Freeswitch-users] mod_managed bind problem
In-Reply-To: <loom.20100520T070859-619@post.gmane.org>
References: <AANLkTilO75HysCaRkSu6oOmorPsItLhojiEjjRqpqdPW@mail.gmail.com>
	<loom.20100520T070859-619@post.gmane.org>
Message-ID: <4BF52858.5020704@puzzled.xs4all.nl>

On 05/20/2010 07:11 AM, babak yakhchali wrote:
> Hi again
> After debuggning freeswitch i found out the problem is that nod_index is not
> initialized and
> if (node_index<= SWITCH_EVENT_ALL&&
> 		switch_event_bind_removable(__FILE__, event_id, subclass_name,
> event_handler, this,&enodes[node_index]) == SWITCH_STATUS_SUCCESS) {
> 		switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "bound to
> %s %s\n", event_name, switch_str_nil(subclass_name));
> 	
> the first part of if condition is always evaluated to false node_index<=
> SWITCH_EVENT_ALL
> when I initialize nod_index to 0 in event consumer constructor it seems working

If this is a bug please report it at http://jira.freeswitch.org 
preferably with a patch against latest git master.

Regards,
Patrick


From talk2ram at gmail.com  Thu May 20 08:38:23 2010
From: talk2ram at gmail.com (ram)
Date: Thu, 20 May 2010 21:08:23 +0530
Subject: [Freeswitch-users] Billing app
In-Reply-To: <1492625819-1274322738-cardhu_decombobulator_blackberry.rim.net-320411259-@bda057.bisx.prodap.on.blackberry>
References: <1492625819-1274322738-cardhu_decombobulator_blackberry.rim.net-320411259-@bda057.bisx.prodap.on.blackberry>
Message-ID: <AANLkTinHvTIXJhLo_aX9U8fULXqqNfqKFJ1gkNTkiLAi@mail.gmail.com>

astpp

On Thu, May 20, 2010 at 8:02 AM, Budi wibowo <bwibowo at gmail.com> wrote:

> Hi guys
> Any suggestion on good billing platform for FS, minimum for postpaid, but
> prepaid also ok.
>
> Thx
> Budi wibowo
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From garrison at codefix.net  Thu May 20 09:44:50 2010
From: garrison at codefix.net (Garrison Hoffman)
Date: Thu, 20 May 2010 12:44:50 -0400
Subject: [Freeswitch-users] Follow Up: Recommendations for adapter
	replacement.
In-Reply-To: <14E11787-9AE2-4D06-A89C-EED6D0F3E55F@gmail.com>
References: <1273761473.1407.26.camel@strontium>
	<14E11787-9AE2-4D06-A89C-EED6D0F3E55F@gmail.com>
Message-ID: <4BF56702.4050404@codefix.net>

First let me thank all the respondents from my original related post,
here is some follow up information which may be useful for anyone
considering similar choices.

I went with a Patton SL4022 which arrived defective, Patton's response
was slow but they seem willing to replace the unit. I'm currently trying 
to see if the vendor will allow me to exchange the unit for a similar 
one made by Audiocodes. I got the impression that Audiocodes has better 
product support, which is something I weight heavily in my evaluations.

I'd like to know if anyone has an experience with these manufacturer's 
product support he'd like to share, but I wouldn't consider other makers 
off topic-- this surely won't be my last VoIP purchase.

-gh


From kris at kriskinc.com  Thu May 20 11:13:51 2010
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Thu, 20 May 2010 14:13:51 -0400
Subject: [Freeswitch-users] mod_dptools/redirect and 300 Multiple Choices
In-Reply-To: <52CD42D7-529A-46D1-AADE-A1E58CB1F881@freeswitch.org>
References: <AANLkTik7DAQjIA6XvpekW_9aSBPhDozvtWTiR-BGkJb3@mail.gmail.com>
	<ADC2C102-DB14-407B-A87F-04F054F96D9B@freeswitch.org>
	<AANLkTikTtQfGNO_nKPbj8Ay3nmY7zbLKCOR996BiftzW@mail.gmail.com>
	<DC75CF0A-17C5-48A5-AD9D-8BBC9D78BFD7@freeswitch.org>
	<AANLkTilLdN-2DkmPBkccMB8B60V-X2Cq0olPDssiyDRx@mail.gmail.com>
	<AANLkTimGjc21-LJssGi50gBC8v1ON5a2UKrKkuBpWuwF@mail.gmail.com>
	<773BA751-D20F-4A04-A8FA-6598656FCF13@freeswitch.org>
	<AANLkTinvFSad8k0GQcCIQw9XQ435KqIiKUNu1uBc0Dp4@mail.gmail.com>
	<52CD42D7-529A-46D1-AADE-A1E58CB1F881@freeswitch.org>
Message-ID: <AANLkTim5u9sN3_Ly0B9Z84GFgVTDiz89EzENb4BG9NhM@mail.gmail.com>

Bringing this up again...

It appears at least one of my platforms does not like multiple Contact
headers or compact headers and I'm not sure how much I can do about
it.  Would it be possible to get FS to send multiple contacts using
one Contact header without using compact headers?

SIP!

On Wed, May 19, 2010 at 6:53 PM, Brian West <brian at freeswitch.org> wrote:
> Its compact.. compressed... oh well long day.
>
> /b
>

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From rupa at rupa.com  Thu May 20 11:23:18 2010
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 20 May 2010 13:23:18 -0500
Subject: [Freeswitch-users] nibblebill never hangs up
In-Reply-To: <AANLkTik1VCDqTy4sFNisPO2HWyKcqBoDNYHdV43FXV2M@mail.gmail.com>
References: <AANLkTik1VCDqTy4sFNisPO2HWyKcqBoDNYHdV43FXV2M@mail.gmail.com>
Message-ID: <AANLkTimWLyz1G9RRvbWsro8T__kR35YqAaZigZdY1PaT@mail.gmail.com>

This one is a bit tricky, I did some research to figure out whats up.

Basically, you can't set the nibble_* vars in the same extension as the call
to bridge due to the order in which the module is called.  The set happens
too late for mod_nibblebill to act on.

You have two options:

Use inline=true on the set -- this causes the var to be set early enough
that mod_nibblebill can act on it
or
after setting transfer to an extension that does the bridge.

Or you can export hte vars and use b-leg nibble billing. but that is another
story.

On Thu, May 20, 2010 at 5:53 AM, Shrouk Khan <shroukkhan at softverk.is> wrote:

> finally in the dial plan :
>
> <extension name="ToIcelandPBX.58d2" >
>   <condition field="destination_number" expression="^(58.*)">
>     <action application="set" data="$${nibble_account}"/>
>   </condition>
>

This set is uselss.  but it is already set by the directory so...  no
biggie.


>   <condition field="destination_number" expression="^(5807)$">
>     <action application="set" data="nibble_rate=30"/>
>

This set is evaluated too late for nibblebill to hook the session.  Change
to:
    <action application="set" data="nibble_rate=30" inline="true"/>


>     <action application="bridge" data="sofia/gateway/ToIcelandPBX/$1"/>
>   </condition>
>
>   <condition field="destination_number" expression="^(5812)$">
>     <action application="set" data="nibble_rate=15"/>
>
same

>     <action application="bridge" data="sofia/gateway/ToIcelandPBX/$1"/>
>   </condition>
>
> </extension>
>
>
> right now the Cash balance for the extension is -192.00 and the extension
> can still make calls jsut fine.
> Am I missing something here ?
>
> --
> Regards
>
> Shrouk Khan (Khan)
>
>
>
-- 
-Rupa
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From anthony.minessale at gmail.com  Thu May 20 12:23:34 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 20 May 2010 14:23:34 -0500
Subject: [Freeswitch-users] mod_dptools/redirect and 300 Multiple Choices
In-Reply-To: <AANLkTim5u9sN3_Ly0B9Z84GFgVTDiz89EzENb4BG9NhM@mail.gmail.com>
References: <AANLkTik7DAQjIA6XvpekW_9aSBPhDozvtWTiR-BGkJb3@mail.gmail.com>
	<ADC2C102-DB14-407B-A87F-04F054F96D9B@freeswitch.org>
	<AANLkTikTtQfGNO_nKPbj8Ay3nmY7zbLKCOR996BiftzW@mail.gmail.com>
	<DC75CF0A-17C5-48A5-AD9D-8BBC9D78BFD7@freeswitch.org>
	<AANLkTilLdN-2DkmPBkccMB8B60V-X2Cq0olPDssiyDRx@mail.gmail.com>
	<AANLkTimGjc21-LJssGi50gBC8v1ON5a2UKrKkuBpWuwF@mail.gmail.com>
	<773BA751-D20F-4A04-A8FA-6598656FCF13@freeswitch.org>
	<AANLkTinvFSad8k0GQcCIQw9XQ435KqIiKUNu1uBc0Dp4@mail.gmail.com>
	<52CD42D7-529A-46D1-AADE-A1E58CB1F881@freeswitch.org>
	<AANLkTim5u9sN3_Ly0B9Z84GFgVTDiz89EzENb4BG9NhM@mail.gmail.com>
Message-ID: <AANLkTimavM46-e2NuI_X4tjK7btb7Nz-AosAckKf2Cqz@mail.gmail.com>

not without intervention from the sofia devs,

Sofia sip does that deep down and we don't have access to it.
even when we supply it as a one big string it renders it out depending on
those settings.


On Thu, May 20, 2010 at 1:13 PM, Kristian Kielhofner <kris at kriskinc.com>wrote:

> Bringing this up again...
>
> It appears at least one of my platforms does not like multiple Contact
> headers or compact headers and I'm not sure how much I can do about
> it.  Would it be possible to get FS to send multiple contacts using
> one Contact header without using compact headers?
>
> SIP!
>
> On Wed, May 19, 2010 at 6:53 PM, Brian West <brian at freeswitch.org> wrote:
> > Its compact.. compressed... oh well long day.
> >
> > /b
> >
>
> --
> Kristian Kielhofner
> http://www.astlinux.org
> http://blog.krisk.org
> http://www.star2star.com
> http://www.submityoursip.com
> http://www.voalte.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <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
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From kris at kriskinc.com  Thu May 20 13:27:29 2010
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Thu, 20 May 2010 16:27:29 -0400
Subject: [Freeswitch-users] mod_dptools/redirect and 300 Multiple Choices
In-Reply-To: <AANLkTimavM46-e2NuI_X4tjK7btb7Nz-AosAckKf2Cqz@mail.gmail.com>
References: <AANLkTik7DAQjIA6XvpekW_9aSBPhDozvtWTiR-BGkJb3@mail.gmail.com>
	<AANLkTikTtQfGNO_nKPbj8Ay3nmY7zbLKCOR996BiftzW@mail.gmail.com>
	<DC75CF0A-17C5-48A5-AD9D-8BBC9D78BFD7@freeswitch.org>
	<AANLkTilLdN-2DkmPBkccMB8B60V-X2Cq0olPDssiyDRx@mail.gmail.com>
	<AANLkTimGjc21-LJssGi50gBC8v1ON5a2UKrKkuBpWuwF@mail.gmail.com>
	<773BA751-D20F-4A04-A8FA-6598656FCF13@freeswitch.org>
	<AANLkTinvFSad8k0GQcCIQw9XQ435KqIiKUNu1uBc0Dp4@mail.gmail.com>
	<52CD42D7-529A-46D1-AADE-A1E58CB1F881@freeswitch.org>
	<AANLkTim5u9sN3_Ly0B9Z84GFgVTDiz89EzENb4BG9NhM@mail.gmail.com>
	<AANLkTimavM46-e2NuI_X4tjK7btb7Nz-AosAckKf2Cqz@mail.gmail.com>
Message-ID: <AANLkTil5cGq6U1C5hvnUgVpBk74rDOKi-J9mDk1mfrpL@mail.gmail.com>

Understood.  I'll see what I can do.

Thanks!

On Thu, May 20, 2010 at 3:23 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> not without intervention from the sofia devs,
> Sofia sip does that deep down and we don't have access to it.
> even when we supply it as a one big string it renders it out depending on
> those settings.
>

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


From marketing at cluecon.com  Thu May 20 16:28:38 2010
From: marketing at cluecon.com (Michael Collins)
Date: Thu, 20 May 2010 16:28:38 -0700
Subject: [Freeswitch-users] Get Registered For ClueCon - You Could Win An
	iPad or MacBook Pro!
Message-ID: <AANLkTinn3esNfNTxytKJZ81gzYoujL4V7acOMFx-eRqE@mail.gmail.com>

ClueCon is coming up fast! We are gearing up for the best ClueCon event yet.
New sponsors, new speakers, and new prizes! This year we are giving away not
one, but TWO Apple 16GB Wifi iPads! Additionally, there will be a 13"
MacBook Pro to be given away. As we did last year, we will be laser
engraving <http://cluecon.com/node/38> the ClueCon sponsors' logos on the
back of the devices. If your organization wants to sponsor this year then
contact us immediately so that you don't miss the deadline for having your
logo included.

ClueCon MMX is being held at the incredibly luxurious Trump Tower
Hotel<http://cluecon.com/hotel-details>in downtown Chicago. Be sure to
get registered right away - rooms are
filling up quickly. You definitely don't want to miss this year's event.

Call 877.742.CLUE to get registered. Remember, those who attended ClueCon
2009 qualify for a $150 discount off of the regular $699 conference price.
That's right - those who attended last year can register again this year for
only $549! Additionally, those who attend the FreeSWITCH
training<http://freeswitchtraining.eventbrite.com/>in San Francisco,
June 28-30, qualify for a special ClueCon rate of $500!
Call today and take advantage of these great deals!

Looking forward to seeing you in Chicago this August,
The ClueCon Team
http://www.cluecon.com
877.742.CLUE
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From math.parent at gmail.com  Thu May 20 17:54:48 2010
From: math.parent at gmail.com (Mathieu Parent)
Date: Fri, 21 May 2010 02:54:48 +0200
Subject: [Freeswitch-users] Mod_skinny: Call for testing!
Message-ID: <AANLkTik8hHRtOci8qTXmZVHOWvVxRg-OPm5zyRAMrWEe@mail.gmail.com>

Hello,

Skinny Call Control Protocol (SCCP) is a VoIP protocol used on several
phones including Cisco IP Phones and Nokia Call Connect 2.0. Contrary
to SIP, the phone is controlled by the server. This allow powerfull
and funny things (like controlling leds, display, ring type, ...) ;
this also allow to use some phones with crappy SIP images, like (guess
what?) Cisco ones.

Since February, mod_skinny is in freeswitch trunk. Many things
happened since. Today the module has reached several milestone (it
compiles on windows, call info is shown early, state handling changed
to a more FreeSWITCH manner, ...).

Most of basic features and more are implemented (from Wiki):
    * Incoming and outgoing calls
    * Shared lines and multiple calls per lines
    * CallInfo set as early as possible
    * New call made from: OffHook, Softkeys (Redial, NewCall),
Stimulus (LastNumberRedial, VoiceMail, SpeedDial)
    * Answering calls via: OffHook, Softkeys (Answer)
    * Holding/Unholding calls via Softkeys (Hold, Resume) or Stimulus
(Hold), with MOH (music on hold)
    * Transfer via Softkeys (Transfer) or Stimulus (Transfer)
    * Ending call via: OnHook, Softkeys (EndCall)
    * Misc: SoftKeys (Backspace)
    * Handle firmware version request (VersionReqMessage) per device
type or per device
    * Patterns for dialed numbers to process (patterns-dialplan and
patterns-context)

Some important missing ones:
* Message Waiting indicator (MWI)
* Bypass Media (help needed)
* Codec Negotiation (help needed)
* Forward (transfer already implemented)
* Complete list:
<http://wiki.freeswitch.org/wiki/Skinny_Call_Control_Protocol#TODO>

How can you help:
* test if it compiles on your platform
* test if it works with your SCCP phone (only 7960 and 7961 have been
tested yet)
* if you don't have an SCCP phone, test using "perl
src/mod/endpoints/mod_skinny/test-skinny.pl" (you need a recent perl,
aka 5.10.1, 5.10.0 has buggy thread implementation, at least on Debian
Lenny).
* write docs about configuration
* propose patches (some are easy like API Commands or events ; some
are harder ;)
* review the code (quality, security, ...)
* If you own a Cisco Communication Manager and you know a feature that
is not in mod_skinny, make network captures!

Some links:
* Instructions are at: http://wiki.freeswitch.org/wiki/Mod_skinny
* Bugs should go here: http://jira.freeswitch.org/browse/MODSKINNY
* Development stuff is at:
http://wiki.freeswitch.org/wiki/Skinny_Call_Control_Protocol
* Suggested features should go here:
http://wiki.freeswitch.org/wiki/Skinny_Call_Control_Protocol#TODO (or
bounty or JIRA)

Any help is appreciated.

happy testing/hacking ;)

Mathieu Parent


From codeghar at gmail.com  Thu May 20 20:25:41 2010
From: codeghar at gmail.com (Code Ghar)
Date: Thu, 20 May 2010 22:25:41 -0500
Subject: [Freeswitch-users] CDRs for Gateways
Message-ID: <AANLkTinISbhS2N7AP-OLA5UskkseP3qGg0Nn-VTTAJWg@mail.gmail.com>

Using mod_cdr_csv, it is easy to create CDRs for registered users. However,
how can we get CDRs for external gateways? For example, FreeSWITCH is
configured to receive calls from one gateway and send them out through
another. In this scenario, when neither inbound nor outbound gateway
registers itself to FreeSWITCH, how can we generate and store CDRs for
inbound gateway?

Similarly, let's say a SIP provider sends a call meant for a registered
user. For example, if a cell phone calls 6175550000, it is routed through
the telephone network to the SIP provider. This provider routes the call to
FreeSWITCH, which looks up in its dial plan that this number should be
routed to extension (or registered user) 1000. There are no CDRs generated
for either the SIP provider's gateway or the registered user in this case.

In both these scenarios, if a call is inbound to FreeSWITCH but has not been
generated by a registered user, mod_cdr_csv does not create CDRs. I have
even tried to set the following parameter in cdr_csv.conf.xml      but it
didn't help.

<param name="legs" value="ab"/>

My question is this: can mod_cdr_csv generate CDRs for inbound calls from
sources other than registered users? If it can't, how can this be achieved?
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From talk2ram at gmail.com  Thu May 20 22:00:53 2010
From: talk2ram at gmail.com (ram)
Date: Fri, 21 May 2010 10:30:53 +0530
Subject: [Freeswitch-users] Gateway Registration Issues
In-Reply-To: <983B7F89-A86E-4AFE-91E6-553D1F0B6495@gmail.com>
References: <AANLkTim-d2_nD3lLS3YlqciwrgJpGP5jMyCfSWtxI_Ly@mail.gmail.com>
	<983B7F89-A86E-4AFE-91E6-553D1F0B6495@gmail.com>
Message-ID: <AANLkTil3UJKkxDU1XHt6FNIpfccng5ynLSnhjMq2kgq_@mail.gmail.com>

thats good suggstion

iam working on same problem

<gateway name="mydomain">
      <param name="username" value="2000"/>
      <param name="auth-username" value="2000"/>
      <param name="password" value="XXXX"/>
      <param name="from-user" value="2000"/>
      <param name="from-domain" value="sip5.domain.com"/>
      <param name="proxy" value="sip5.domain.com"/>
      <param name="register-proxy" value="sip5.domain.com"/>
      <param name="outbound-proxy" value="sip5.domain.com"/>
      <param name="expire-seconds" value="800"/>
      <param name="register" value="true"/>
      <param name="register-transport" value="udp"/>
      <param name="retry-seconds" value="60"/>
      <param name="ping" value="6000"/>
      <param name="context" value="public"/>
      <param name="caller-id-in-from" value="true"/>
      <param name="extension_in_contact" value="true"/>
    </gateway>
</include>


and My gateway side still show like below



AOR:: 2000 at sip5.otherdomain.com
                Contact::
sip:gw+mydomain at x.x.x.x:5080;transport=udp;gw=mydomain
Q=
                        Expires:: 794
                        Callid:: 30b6aa56-6491-11df-86f8-797747db92ac
                        Cseq:: 131089695
                        User-agent:: FreeSWITCH-mod_sofia/1.0.head-svn-17503
                        State:: CS_NEW
                        Flags:: 0
                        Cflag:: 0
                        Socket:: udp:x.x.x.x:5060
                        Methods:: 5631


On Thu, May 13, 2010 at 6:28 PM, David Ponzone <david.ponzone at gmail.com>wrote:

> Aloysius,,
>
> That is an old-skool carrier.
> You have to add this line to your gateway params:
>
> <param name="extension-in-contact" value="true"/>
>
>
>   David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
>  tel:      0811 46 26 26
>  www.ipeva.fr  -   www.ipeva-studio.com
>
>  *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
>  Le 13/05/2010 ? 14:46, Aloysius Lloyd a ?crit :
>
>   Hi All,
>
> I am trying to connect IRISTEL ( CLEC ) in Canada and FreeSWITCH. I have
> the following gateway configuration
>
>  <include>
>   <gateway name="wwc_test">
>     <param name="auth-username" value="username"/>
>     <param name="username" value="14161231234" />
>     <param name="password" value="password" />
>     <param name="proxy" value="sbc1.iristel.net" />
>     <param name="realm" value="iristel.net" />
>     <param name="expire-seconds" value="120"/>
>     <param name="register" value="true"/>
>     <param name="retry-seconds" value="30"/>
>     <param name="ping" value="25"/>
>   </gateway>
> </include>
>
>
> The above configuration is not working. I have the following error in cli
> Error.
>
>  *2010-05-13 08:34:15.987465 [ERR] sofia_reg.c:1501 14161231234
> Registration Failed with status Conflict [409]. failure #1*
>
> ====
>
> Also carrier saying there is conflict in the contact field.Here is how
> they receive now.
>
>  *Contact: <
> sip:gw+test_iristel at 10.20.30.1:5080;transport=udp;gw=test_iristel>.*
>
> But the contact field should be like below
>
> Contact: <14161231234 at 10.20.30.1:5080;
>
> Any suggestions?
>
> Thanks in advance.
>
> Lloyd
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From the_spide21 at yahoo.com  Thu May 20 19:37:06 2010
From: the_spide21 at yahoo.com (Carlos Lopez)
Date: Thu, 20 May 2010 19:37:06 -0700 (PDT)
Subject: [Freeswitch-users] Voip fail over device solutions, please help
Message-ID: <327595.9479.qm@web45107.mail.sp1.yahoo.com>

Hi all,

I am planning to deploy some test over a voip server, I'll be setting up two voip servers, but I want to know if there are devices that can take one T1/E1 PSTN lease and then transform that signal into IP and connect it to a ethernet card on each voip server. Also I'd like to know if I can do the same with the FXS analog phones which will be connected to a telphone patch panel and from that patch panel to a TDM 50pins PCI card, for example:

1-
phone1<--->||Telephone  ||<--Amphenolcable-->|Failover|<--->*1
phone2<--->||patch panel||                   |device..|<--->*2

2-
(*1-VOIP_SRV1)<--IP-->|Ethernet|<--TDMoE--->|Failover|<--->T1/E1
(*2-VOIP_SRV2)<--IP-->|Switch  |            |device  |

What are the best IP to analog phones PCI cards supported by Freeswitch, any hints?.

Thanks in advanced for your help.

Carlos.



      



From david.ponzone at gmail.com  Fri May 21 00:03:39 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Fri, 21 May 2010 09:03:39 +0200
Subject: [Freeswitch-users] CDRs for Gateways
In-Reply-To: <AANLkTinISbhS2N7AP-OLA5UskkseP3qGg0Nn-VTTAJWg@mail.gmail.com>
References: <AANLkTinISbhS2N7AP-OLA5UskkseP3qGg0Nn-VTTAJWg@mail.gmail.com>
Message-ID: <F10C2176-E9E9-476D-BF0B-40AB05EDFE5A@gmail.com>

Code,

I think you have a major issue or configuration error with your FS  
setup.
FS, as any B2BUA, generates a CDR for any inbound leg, either from a  
registered user or from a non-registered gateway.
It doesn't care where it comes from, it just does it.
I can barely imagine a softswitch that would not write the CDR  
matching a call that was routed through it.

The CDR for the outbound leg is optional.

Can you confirm you are looking in Master.csv ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 21/05/2010 ? 05:25, Code Ghar a ?crit :

> Using mod_cdr_csv, it is easy to create CDRs for registered users.  
> However, how can we get CDRs for external gateways? For example,  
> FreeSWITCH is configured to receive calls from one gateway and send  
> them out through another. In this scenario, when neither inbound nor  
> outbound gateway registers itself to FreeSWITCH, how can we generate  
> and store CDRs for inbound gateway?
>
> Similarly, let's say a SIP provider sends a call meant for a  
> registered user. For example, if a cell phone calls 6175550000, it  
> is routed through the telephone network to the SIP provider. This  
> provider routes the call to FreeSWITCH, which looks up in its dial  
> plan that this number should be routed to extension (or registered  
> user) 1000. There are no CDRs generated for either the SIP  
> provider's gateway or the registered user in this case.
>
> In both these scenarios, if a call is inbound to FreeSWITCH but has  
> not been generated by a registered user, mod_cdr_csv does not create  
> CDRs. I have even tried to set the following parameter in  
> cdr_csv.conf.xml      but it didn't help.
>
> <param name="legs" value="ab"/>
>
> My question is this: can mod_cdr_csv generate CDRs for inbound calls  
> from sources other than registered users? If it can't, how can this  
> be achieved?
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From david.ponzone at gmail.com  Fri May 21 00:05:21 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Fri, 21 May 2010 09:05:21 +0200
Subject: [Freeswitch-users] Gateway Registration Issues
In-Reply-To: <AANLkTil3UJKkxDU1XHt6FNIpfccng5ynLSnhjMq2kgq_@mail.gmail.com>
References: <AANLkTim-d2_nD3lLS3YlqciwrgJpGP5jMyCfSWtxI_Ly@mail.gmail.com>
	<983B7F89-A86E-4AFE-91E6-553D1F0B6495@gmail.com>
	<AANLkTil3UJKkxDU1XHt6FNIpfccng5ynLSnhjMq2kgq_@mail.gmail.com>
Message-ID: <4F8EA7B9-B642-4C66-AF5A-ADDF554DA166@gmail.com>

You restarted the gateway ?

sofia profile external killgw mydomain
sofia profile external rescan reloadxml

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 21/05/2010 ? 07:00, ram a ?crit :

>
>
>
> thats good suggstion
>
> iam working on same problem
>
> <gateway name="mydomain">
>       <param name="username" value="2000"/>
>       <param name="auth-username" value="2000"/>
>       <param name="password" value="XXXX"/>
>       <param name="from-user" value="2000"/>
>       <param name="from-domain" value="sip5.domain.com"/>
>       <param name="proxy" value="sip5.domain.com"/>
>       <param name="register-proxy" value="sip5.domain.com"/>
>       <param name="outbound-proxy" value="sip5.domain.com"/>
>       <param name="expire-seconds" value="800"/>
>       <param name="register" value="true"/>
>       <param name="register-transport" value="udp"/>
>       <param name="retry-seconds" value="60"/>
>       <param name="ping" value="6000"/>
>       <param name="context" value="public"/>
>       <param name="caller-id-in-from" value="true"/>
>       <param name="extension_in_contact" value="true"/>
>     </gateway>
> </include>
>
>
> and My gateway side still show like below
>
>
>
> AOR:: 2000 at sip5.otherdomain.com
>                 Contact:: sip:gw+mydomain at x.x.x.x:5080;transport=udp;gw=mydomain 
>  Q=
>                         Expires:: 794
>                         Callid:: 30b6aa56-6491-11df-86f8-797747db92ac
>                         Cseq:: 131089695
>                         User-agent:: FreeSWITCH-mod_sofia/1.0.head- 
> svn-17503
>                         State:: CS_NEW
>                         Flags:: 0
>                         Cflag:: 0
>                         Socket:: udp:x.x.x.x:5060
>                         Methods:: 5631
>
>
> On Thu, May 13, 2010 at 6:28 PM, David Ponzone <david.ponzone at gmail.com 
> > wrote:
> Aloysius,,
>
> That is an old-skool carrier.
> You have to add this line to your gateway params:
>
> <param name="extension-in-contact" value="true"/>
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et  
> ?tablis ? l'intention exclusive de ses destinataires. Toute  
> utilisation ou diffusion non autoris?e est interdite. Tout message  
> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou  
> falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le  
> d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
>
> Le 13/05/2010 ? 14:46, Aloysius Lloyd a ?crit :
>
>> Hi All,
>>
>> I am trying to connect IRISTEL ( CLEC ) in Canada and FreeSWITCH. I  
>> have the following gateway configuration
>>
>> <include>
>>   <gateway name="wwc_test">
>>     <param name="auth-username" value="username"/>
>>     <param name="username" value="14161231234" />
>>     <param name="password" value="password" />
>>     <param name="proxy" value="sbc1.iristel.net" />
>>     <param name="realm" value="iristel.net" />
>>     <param name="expire-seconds" value="120"/>
>>     <param name="register" value="true"/>
>>     <param name="retry-seconds" value="30"/>
>>     <param name="ping" value="25"/>
>>   </gateway>
>> </include>
>>
>>
>> The above configuration is not working. I have the following error  
>> in cli Error.
>>
>> 2010-05-13 08:34:15.987465 [ERR] sofia_reg.c:1501 14161231234  
>> Registration Failed with status Conflict [409]. failure #1
>>
>> ====
>>
>> Also carrier saying there is conflict in the contact field.Here is  
>> how they receive now.
>>
>> Contact: <sip:gw+test_iristel at 10.20.30.1:5080;transport=udp;gw=test_iristel 
>> >.
>>
>> But the contact field should be like below
>>
>> Contact: <14161231234 at 10.20.30.1:5080;
>>
>> Any suggestions?
>>
>> Thanks in advance.
>>
>> Lloyd
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From david.ponzone at gmail.com  Fri May 21 00:09:51 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Fri, 21 May 2010 09:09:51 +0200
Subject: [Freeswitch-users] Voip fail over device solutions, please help
In-Reply-To: <327595.9479.qm@web45107.mail.sp1.yahoo.com>
References: <327595.9479.qm@web45107.mail.sp1.yahoo.com>
Message-ID: <AFA843D7-CFD5-4AB2-A63B-45F6ED63233A@gmail.com>

Carlos,

if what you want to achieve is to use a T1/E1 line that is remote, so  
not connected physically to your FS box, I think you need this:

http://support.red-fone.com/index.php?_m=knowledgebase&_a=viewarticle&kbarticleid=17

There could be other vendors, I have absolutely no knowledge about  
this, I just found out about it recently.
I don't even know if it works.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 21/05/2010 ? 04:37, Carlos Lopez a ?crit :

> Hi all,
>
> I am planning to deploy some test over a voip server, I'll be  
> setting up two voip servers, but I want to know if there are devices  
> that can take one T1/E1 PSTN lease and then transform that signal  
> into IP and connect it to a ethernet card on each voip server. Also  
> I'd like to know if I can do the same with the FXS analog phones  
> which will be connected to a telphone patch panel and from that  
> patch panel to a TDM 50pins PCI card, for example:
>
> 1-
> phone1<--->||Telephone  ||<--Amphenolcable-->|Failover|<--->*1
> phone2<--->||patch panel||                   |device..|<--->*2
>
> 2-
> (*1-VOIP_SRV1)<--IP-->|Ethernet|<--TDMoE--->|Failover|<--->T1/E1
> (*2-VOIP_SRV2)<--IP-->|Switch  |            |device  |
>
> What are the best IP to analog phones PCI cards supported by  
> Freeswitch, any hints?.
>
> Thanks in advanced for your help.
>
> Carlos.
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mbrancaleoni at voismart.it  Fri May 21 03:17:00 2010
From: mbrancaleoni at voismart.it (Matteo)
Date: Fri, 21 May 2010 12:17:00 +0200 (CEST)
Subject: [Freeswitch-users] Voip fail over device solutions, please help
In-Reply-To: <327595.9479.qm@web45107.mail.sp1.yahoo.com>
Message-ID: <1417628203.1176.1274437020032.JavaMail.root@mx.voismart.com>

Hi,

----- "Carlos Lopez" <the_spide21 at yahoo.com> ha scritto:

> Hi all,
> 
> I am planning to deploy some test over a voip server, I'll be setting
> up two voip servers, but I want to know if there are devices that can
> take one T1/E1 PSTN lease and then transform that signal into IP and
> connect it to a ethernet card on each voip server.

you can setup a pri card into the boxes and use isdnguard (from junghanns)
or berofos (from beronet) to failover the main pri line to one of the boxes.

for analog there're some similar solutions, but only for small ports
numebers...

mat


From romerocarlos24 at gmail.com  Thu May 20 23:46:06 2010
From: romerocarlos24 at gmail.com (Carlos Romero)
Date: Fri, 21 May 2010 02:46:06 -0400
Subject: [Freeswitch-users]  mod_dptools/redirect and 300 Multiple Choices
Message-ID: <AANLkTinYtV2CL1H_hME51kP4OpJLaYeuQK0jUEnrimAz@mail.gmail.com>

what do you mean that you don't have access to it?

you have the code

On Thu, May 20, 2010 at 3:23 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> not without intervention from the sofia devs,
> Sofia sip does that deep down and we don't have access to it.
> even when we supply it as a one big string it renders it out depending on
> those settings.
>


From mike at jerris.com  Fri May 21 06:39:50 2010
From: mike at jerris.com (Michael Jerris)
Date: Fri, 21 May 2010 09:39:50 -0400
Subject: [Freeswitch-users] mod_dptools/redirect and 300 Multiple Choices
In-Reply-To: <AANLkTinYtV2CL1H_hME51kP4OpJLaYeuQK0jUEnrimAz@mail.gmail.com>
References: <AANLkTinYtV2CL1H_hME51kP4OpJLaYeuQK0jUEnrimAz@mail.gmail.com>
Message-ID: <E1810BD2-9222-4AB8-BF5E-4094282A843D@jerris.com>

Thank you for your generous offer to supply this patch.  I look forward to seeing it contributed.

Mike

On May 21, 2010, at 2:46 AM, Carlos Romero wrote:

> what do you mean that you don't have access to it?
> 
> you have the code
> 
> On Thu, May 20, 2010 at 3:23 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> not without intervention from the sofia devs,
>> Sofia sip does that deep down and we don't have access to it.
>> even when we supply it as a one big string it renders it out depending on
>> those settings.
>> 



From stevendt at primrosebank.net  Fri May 21 06:46:41 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Fri, 21 May 2010 14:46:41 +0100
Subject: [Freeswitch-users] Mod_skinny: Call for testing!
References: <AANLkTik8hHRtOci8qTXmZVHOWvVxRg-OPm5zyRAMrWEe@mail.gmail.com>
Message-ID: <F32D0E59CC994113BC88C7902AFC87BF@bp1.ad.bp.com>

Mathieu,

just updated to 17539 (SVN) and skinny is not included in the Windows Build 
Solution to VS2008

Does it need to be compiled & linked outside of the VS Build ?

regards
Dave



----- Original Message ----- 
From: "Mathieu Parent" <math.parent at gmail.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Friday, May 21, 2010 1:54 AM
Subject: [Freeswitch-users] Mod_skinny: Call for testing!


> Hello,
>
> Skinny Call Control Protocol (SCCP) is a VoIP protocol used on several
> phones including Cisco IP Phones and Nokia Call Connect 2.0. Contrary
> to SIP, the phone is controlled by the server. This allow powerfull
> and funny things (like controlling leds, display, ring type, ...) ;
> this also allow to use some phones with crappy SIP images, like (guess
> what?) Cisco ones.
>
> Since February, mod_skinny is in freeswitch trunk. Many things
> happened since. Today the module has reached several milestone (it
> compiles on windows, call info is shown early, state handling changed
> to a more FreeSWITCH manner, ...).
>
> Most of basic features and more are implemented (from Wiki):
>    * Incoming and outgoing calls
>    * Shared lines and multiple calls per lines
>    * CallInfo set as early as possible
>    * New call made from: OffHook, Softkeys (Redial, NewCall),
> Stimulus (LastNumberRedial, VoiceMail, SpeedDial)
>    * Answering calls via: OffHook, Softkeys (Answer)
>    * Holding/Unholding calls via Softkeys (Hold, Resume) or Stimulus
> (Hold), with MOH (music on hold)
>    * Transfer via Softkeys (Transfer) or Stimulus (Transfer)
>    * Ending call via: OnHook, Softkeys (EndCall)
>    * Misc: SoftKeys (Backspace)
>    * Handle firmware version request (VersionReqMessage) per device
> type or per device
>    * Patterns for dialed numbers to process (patterns-dialplan and
> patterns-context)
>
> Some important missing ones:
> * Message Waiting indicator (MWI)
> * Bypass Media (help needed)
> * Codec Negotiation (help needed)
> * Forward (transfer already implemented)
> * Complete list:
> <http://wiki.freeswitch.org/wiki/Skinny_Call_Control_Protocol#TODO>
>
> How can you help:
> * test if it compiles on your platform
> * test if it works with your SCCP phone (only 7960 and 7961 have been
> tested yet)
> * if you don't have an SCCP phone, test using "perl
> src/mod/endpoints/mod_skinny/test-skinny.pl" (you need a recent perl,
> aka 5.10.1, 5.10.0 has buggy thread implementation, at least on Debian
> Lenny).
> * write docs about configuration
> * propose patches (some are easy like API Commands or events ; some
> are harder ;)
> * review the code (quality, security, ...)
> * If you own a Cisco Communication Manager and you know a feature that
> is not in mod_skinny, make network captures!
>
> Some links:
> * Instructions are at: http://wiki.freeswitch.org/wiki/Mod_skinny
> * Bugs should go here: http://jira.freeswitch.org/browse/MODSKINNY
> * Development stuff is at:
> http://wiki.freeswitch.org/wiki/Skinny_Call_Control_Protocol
> * Suggested features should go here:
> http://wiki.freeswitch.org/wiki/Skinny_Call_Control_Protocol#TODO (or
> bounty or JIRA)
>
> Any help is appreciated.
>
> happy testing/hacking ;)
>
> Mathieu Parent
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From fs-list at communicatefreely.net  Fri May 21 06:48:07 2010
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Fri, 21 May 2010 09:48:07 -0400
Subject: [Freeswitch-users] Voip fail over device solutions, please help
In-Reply-To: <1417628203.1176.1274437020032.JavaMail.root@mx.voismart.com>
References: <1417628203.1176.1274437020032.JavaMail.root@mx.voismart.com>
Message-ID: <4BF68F17.8000107@communicatefreely.net>

Looks like other folks have answered about PRI.

You can in fact fail over large numbers of analog lines.  Look for a power fail relay type device.
I know that Dees Telecom makes some of these.  They have 50 pin Centronics connectors on them, and
can switch large numbers of circuits (all at the same time).  Usually, these are triggered by loss
of electrical power, so you may have to come up with a creative way to activate this device.  If you
can generate some sort of heartbeat signal on a serial or parallel port, that could be used.
Alternately, there are some serial and Ethernet to contact closure interfaces (Adantech's ADAM line
of industrial automation products) that could control the relays.  You would have to write a script
to generate the MODBUS commands that these use, but the protocol is well documented.  If you used
the Ethernet version, either box could issue a command to "switch to me".  That is a little bit more
expensive than making your own heartbeat detector, but it doesn't require any soldering.  You can
also use these boxes to give you status on things, and remotely reboot equipment.

-Tim

Matteo wrote:
> Hi,
> 
> ----- "Carlos Lopez" <the_spide21 at yahoo.com> ha scritto:
> 
>> Hi all,
>>
>> I am planning to deploy some test over a voip server, I'll be setting
>> up two voip servers, but I want to know if there are devices that can
>> take one T1/E1 PSTN lease and then transform that signal into IP and
>> connect it to a ethernet card on each voip server.
> 
> you can setup a pri card into the boxes and use isdnguard (from junghanns)
> or berofos (from beronet) to failover the main pri line to one of the boxes.
> 
> for analog there're some similar solutions, but only for small ports
> numebers...
> 
> mat
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From frank at impactfax.com  Fri May 21 07:05:30 2010
From: frank at impactfax.com (Frank @ Impact)
Date: Fri, 21 May 2010 10:05:30 -0400
Subject: [Freeswitch-users] control sip_cid_type from sip_profile
Message-ID: <748A4880639A4DD0B1982E5251B6E749@ws4>

for RPID/PAI: sip_cid_type - 
Is there a way to control this setting in the gateway profile instead of
having to do it in the bridge setup? 
Some providers seem to want rpid and others want p-asserted.
 
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From math.parent at gmail.com  Fri May 21 07:24:38 2010
From: math.parent at gmail.com (Mathieu Parent)
Date: Fri, 21 May 2010 16:24:38 +0200
Subject: [Freeswitch-users] Mod_skinny: Call for testing!
In-Reply-To: <F32D0E59CC994113BC88C7902AFC87BF@bp1.ad.bp.com>
References: <AANLkTik8hHRtOci8qTXmZVHOWvVxRg-OPm5zyRAMrWEe@mail.gmail.com>
	<F32D0E59CC994113BC88C7902AFC87BF@bp1.ad.bp.com>
Message-ID: <AANLkTikJ1biYWWY-M4qYMWgsYfGelxvFu91EVnBqDlcq@mail.gmail.com>

On Fri, May 21, 2010 at 3:46 PM, Dave Stevenson
<stevendt at primrosebank.net> wrote:
> Mathieu,
>
> just updated to 17539 (SVN) and skinny is not included in the Windows Build
> Solution to VS2008
>
> Does it need to be compiled & linked outside of the VS Build ?

SVN is out of sync, git has Windows support since yesterday:

commit bfb6a8b1f5f835008d1e0fbd90d2e120814f341a
Author: Mathieu Parent <math.parent at gmail.com>
Date:   Thu May 20 23:19:10 2010 +0200

    Skinny: Initial Windows support

    Thanks to Peter Olsson

    Closes: MODSKINNY-3

I have not been able to test it, you can comment MODSKINNY-3 if you
still have problem after the patch applied.

Mathieu Parent


From testeador01 at gmail.com  Fri May 21 07:45:23 2010
From: testeador01 at gmail.com (Milena)
Date: Fri, 21 May 2010 09:45:23 -0500
Subject: [Freeswitch-users] kamailio and freeswitch integration
In-Reply-To: <4BF19344.4090607@gmail.com>
References: <4BF105DA.2060209@gmail.com>
	<AANLkTin5sPaagA4RglV3JfJ0rTZJy8_FzNwd4IpVx3LJ@mail.gmail.com>
	<780492B9CF324B359CA987710CA7738D@MOBILEE1705>
	<201005171355.17228.sos@sokhapkin.dyndns.org>
	<A3BDB256F37F4E468E8268F529E77081@MOBILEE1705>
	<AANLkTikpv0hVHZUgzCR6HsVuR73rmrUnEnOOSXp3pZq6@mail.gmail.com>
	<500DFBB8420E46DEAE5A58BAFDA5066C@MOBILEE1705>
	<4BF19344.4090607@gmail.com>
Message-ID: <AANLkTilfe4eWrA-Bso94QqfoBow9y97_wwV0lJNtyIEt@mail.gmail.com>

Hello Daniel,

I will find some time and copy the information to the wiki if i can get your
authorization to do so, is it ok? or would you rather have traffic going
into your website? i would post a link to the original article anyways :)

Waiting for your reply,
-Milena

2010/5/17 Daniel-Constantin Mierla <miconda at gmail.com>

>  Hello,
>
> sorry for inconveniences, but dns propagation really sucks big this time --
> record is more than one week old and seems some parts of the world still
> have an older cache. Being in travel to sipit26 cannot invest much time in
> it and copy to fs wiki, so i just redirected one of older records to same
> site, use this link for a while:
>
> http://ngs.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
>
> Colon - ':' -  in link is not a problem.
>
> @Phillip Jones - probably you got exactly some restart of the web server
> while redirecting the subdomain, site is up 99.999%.
>
> Cheers,
> Daniel
>
>
>
> On 5/17/10 8:49 PM, Madovsky wrote:
>
> I think the problem is the " : "  in the link....
>
> ----- Original Message -----
> *From:* Phillip Jones <pjintheusa at gmail.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Monday, May 17, 2010 2:22 PM
> *Subject:* Re: [Freeswitch-users] kamailio and freeswitch integration
>
>  I think their server is up and down. Worked for me when I posted. Does not
> now. Base URL ok though.
>
> On Mon, May 17, 2010 at 2:10 PM, Madovsky <infos at madovsky.org> wrote:
>
>> weird...
>> from canada (montreal)
>> not works
>>
>> ----- Original Message -----
>> From: "Sergey Okhapkin" <sos at sokhapkin.dyndns.org>
>> To: <freeswitch-users at lists.freeswitch.org>
>>  Sent: Monday, May 17, 2010 1:55 PM
>> Subject: Re: [Freeswitch-users] kamailio and freeswitch integration
>>
>>
>> > The link works fine to me.
>> >
>> > On Monday 17 May 2010, Madovsky wrote:
>> >> doesn't work
>> >>   ----- Original Message -----
>> >>   From: Phillip Jones
>> >>   To: freeswitch-users at lists.freeswitch.org
>> >>   Sent: Monday, May 17, 2010 1:19 PM
>> >>   Subject: Re: [Freeswitch-users] kamailio and freeswitch integration
>> >>
>> >>
>> >>   For those that cannot wait (like me!)
>> >>
>> >>   http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
>> >>
>> >>
>> >>   On Mon, May 17, 2010 at 12:48 PM, Giovanni Maruzzelli
>> >>  <gmaruzz at celliax.org> wrote:
>> >>
>> >>     Daniel,
>> >>
>> >>     your link seems not reachable.
>> >>     We'll wait for the wiki page :).
>> >>
>> >>     Btw, much appreciated, thanks!
>> >>
>> >>     -giovanni
>> >>
>> >>     On Mon, May 17, 2010 at 4:54 PM, Michael S  Collins
>> >> <msc at freeswitch.org>
>> > wrote:
>> >>     > Definitely! Create a wiki page named Kamailio and add your
>> content
>> >>     > there. We will then link to it from other appropriate pages.
>> >>     >
>> >>     > Thanks!
>> >>     > -MC (IRC:mercutioviz)
>> >>     >
>> >>     > Sent from my iPhone
>> >>     >
>> >>     > On May 17, 2010, at 2:01 AM, Daniel-Constantin Mierla
>> >>     >
>> >>     > <miconda at gmail.com> wrote:
>> >>     >> Hello,
>> >>     >>
>> >>     >> I put together a tutorial about using kamailio (openser) and
>> >>     >> freeswtich
>> >>     >> together: the proxy takes care of authentication and
>> registration,
>> >>     >> freeswitch of media services, here is a link:
>> >>     >>
>> >>     >>
>> >> http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
>> >>     >>
>> >>     >> Is it ok to upload it to FS wiki so others can add to it?
>> >>     >>
>> >>     >> Cheers,
>> >>     >> Daniel
>> >>     >>
>> >>     >> --
>> >>     >> Daniel-Constantin Mierla
>> >>     >> Kamailio (OpenSER) Advanced Training
>> >>     >> Miami, Fl, USA - June 21-23, 2010
>> >>     >> http://www.asipto.com/index.php/kamailio-advanced-training/
>> >>     >>
>> >>     >>
>> >>     >> _______________________________________________
>> >>     >> FreeSWITCH-users mailing list
>> >>     >> FreeSWITCH-users at lists.freeswitch.org
>> >>     >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>     >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> >>     >> users
>> >>     >> http://www.freeswitch.org
>> >>     >
>> >>     > _______________________________________________
>> >>     > FreeSWITCH-users mailing list
>> >>     > FreeSWITCH-users at lists.freeswitch.org
>> >>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>     >
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
>> >>     >ers http://www.freeswitch.org
>> >>
>> >>     --
>> >>     Sincerely,
>> >>
>> >>     Giovanni Maruzzelli
>> >>     Cell : +39-347-2665618
>> >>
>> >>
>> >>     _______________________________________________
>> >>     FreeSWITCH-users mailing list
>> >>     FreeSWITCH-users at lists.freeswitch.org
>> >>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >>  UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>  http://www.freeswitch.org
>> >>
>> >>
>> >>
>> >>
>> >>
>> >>
>> ---------------------------------------------------------------------------
>> >> ---
>> >>
>> >>
>> >>   _______________________________________________
>> >>   FreeSWITCH-users mailing list
>> >>   FreeSWITCH-users at lists.freeswitch.org
>> >>   http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>   http://www.freeswitch.org
>> >>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>  ------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> --
> Daniel-Constantin Mierla
> Kamailio (OpenSER) Advanced Training
> Miami, Fl, USA - June 21-23, 2010
> http://www.asipto.com/index.php/kamailio-advanced-training/
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From peter.olsson at visionutveckling.se  Fri May 21 08:16:34 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 21 May 2010 17:16:34 +0200
Subject: [Freeswitch-users] Mod_skinny: Call for testing!
In-Reply-To: <AANLkTikJ1biYWWY-M4qYMWgsYfGelxvFu91EVnBqDlcq@mail.gmail.com>
References: <AANLkTik8hHRtOci8qTXmZVHOWvVxRg-OPm5zyRAMrWEe@mail.gmail.com>
	<F32D0E59CC994113BC88C7902AFC87BF@bp1.ad.bp.com>,
	<AANLkTikJ1biYWWY-M4qYMWgsYfGelxvFu91EVnBqDlcq@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C557DE57F78@cooper>

I tried building today, and it builds on Windows on latest git - just needs some testing now :)

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Mathieu Parent [math.parent at gmail.com]
Skickat: den 21 maj 2010 16:24
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Mod_skinny: Call for testing!

On Fri, May 21, 2010 at 3:46 PM, Dave Stevenson
<stevendt at primrosebank.net> wrote:
> Mathieu,
>
> just updated to 17539 (SVN) and skinny is not included in the Windows Build
> Solution to VS2008
>
> Does it need to be compiled & linked outside of the VS Build ?

SVN is out of sync, git has Windows support since yesterday:

commit bfb6a8b1f5f835008d1e0fbd90d2e120814f341a
Author: Mathieu Parent <math.parent at gmail.com>
Date:   Thu May 20 23:19:10 2010 +0200

    Skinny: Initial Windows support

    Thanks to Peter Olsson

    Closes: MODSKINNY-3

I have not been able to test it, you can comment MODSKINNY-3 if you
still have problem after the patch applied.

Mathieu Parent

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

!DSPAM:4bf6994c32931601618424!



From pjintheusa at gmail.com  Fri May 21 08:41:44 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Fri, 21 May 2010 11:41:44 -0400
Subject: [Freeswitch-users] Quick question on Originate through api
Message-ID: <AANLkTilUfOj-ttfoo6SePy0bb5q0BB1ARxyExE8L1385@mail.gmail.com>

Hi there,

In my mind - where 6775557767 is a registered user, the following two are
the same:

apiResult = fsApi.Execute("originate",
string.Format("{{ignore_early_media=false,absolute_codec_string='PCMU',origination_caller_id_number=5556667776}}user/6775557767
'

apiResult = fsApi.Execute("originate",
string.Format("{{ignore_early_media=false,absolute_codec_string='PCMU',origination_caller_id_number=5556667776}}sofia/internal/6775557767 at 121.763.76.76'

i.e. user/ and sofia/internal are the same thing.

However on my system the user/ works great but sofia/internal loops call
back through in the public context.

Can someone tell me the difference between these options?

What am I missing?

Thanks


Phil
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From peter.olsson at visionutveckling.se  Fri May 21 08:52:48 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 21 May 2010 17:52:48 +0200
Subject: [Freeswitch-users] Quick question on Originate through api
In-Reply-To: <AANLkTilUfOj-ttfoo6SePy0bb5q0BB1ARxyExE8L1385@mail.gmail.com>
References: <AANLkTilUfOj-ttfoo6SePy0bb5q0BB1ARxyExE8L1385@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C557DE57F7A@cooper>

For sofia/internal/ you should use a % as a separator I think, instead of @-sign.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Phillip Jones [pjintheusa at gmail.com]
Skickat: den 21 maj 2010 17:41
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] Quick question on Originate through api

Hi there,

In my mind - where 6775557767 is a registered user, the following two are the same:

apiResult = fsApi.Execute("originate", string.Format("{{ignore_early_media=false,absolute_codec_string='PCMU',origination_caller_id_number=5556667776}}user/6775557767 '

apiResult = fsApi.Execute("originate", string.Format("{{ignore_early_media=false,absolute_codec_string='PCMU',origination_caller_id_number=5556667776}}sofia/internal/6775557767 at 121.763.76.76 '

i.e. user/ and sofia/internal are the same thing.

However on my system the user/ works great but sofia/internal loops call back through in the public context.

Can someone tell me the difference between these options?

What am I missing?

Thanks


Phil



!DSPAM:4bf6abb932934294689129!


From brian at freeswitch.org  Fri May 21 09:03:07 2010
From: brian at freeswitch.org (Brian West)
Date: Fri, 21 May 2010 11:03:07 -0500
Subject: [Freeswitch-users] Quick question on Originate through api
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C557DE57F7A@cooper>
References: <AANLkTilUfOj-ttfoo6SePy0bb5q0BB1ARxyExE8L1385@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C557DE57F7A@cooper>
Message-ID: <50A3FFB9-E66D-4986-AA01-6F162B38F18D@freeswitch.org>

http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ#Q:_What_is_the_difference_between_using_a_.25_and_.40_in_a_sofia_dial_string.3F

Or at the very least read the FAQ... We wrote it for a reason ;)

/b

On May 21, 2010, at 10:52 AM, Peter Olsson wrote:

> For sofia/internal/ you should use a % as a separator I think, instead of @-sign.
> 
> /Peter
> ________



From infos at madovsky.org  Fri May 21 09:07:17 2010
From: infos at madovsky.org (Madovsky)
Date: Fri, 21 May 2010 12:07:17 -0400
Subject: [Freeswitch-users] Quick question on Originate through api
References: <AANLkTilUfOj-ttfoo6SePy0bb5q0BB1ARxyExE8L1385@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C557DE57F7A@cooper>
Message-ID: <BD50A783B5B646B3AB6ECEFB6EDB23D4@MOBILEE1705>

not necessary ....

----- Original Message ----- 
From: "Peter Olsson" <peter.olsson at visionutveckling.se>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Friday, May 21, 2010 11:52 AM
Subject: Re: [Freeswitch-users] Quick question on Originate through api


For sofia/internal/ you should use a % as a separator I think, instead of 
@-sign.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org 
[freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Phillip Jones 
[pjintheusa at gmail.com]
Skickat: den 21 maj 2010 17:41
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] Quick question on Originate through api

Hi there,

In my mind - where 6775557767 is a registered user, the following two are 
the same:

apiResult = fsApi.Execute("originate", 
string.Format("{{ignore_early_media=false,absolute_codec_string='PCMU',origination_caller_id_number=5556667776}}user/6775557767 
'

apiResult = fsApi.Execute("originate", 
string.Format("{{ignore_early_media=false,absolute_codec_string='PCMU',origination_caller_id_number=5556667776}}sofia/internal/6775557767 at 121.763.76.76 
'

i.e. user/ and sofia/internal are the same thing.

However on my system the user/ works great but sofia/internal loops call 
back through in the public context.

Can someone tell me the difference between these options?

What am I missing?

Thanks


Phil



!DSPAM:4bf6abb932934294689129!

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



From kris at kriskinc.com  Fri May 21 09:40:00 2010
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Fri, 21 May 2010 12:40:00 -0400
Subject: [Freeswitch-users] Quick question on Originate through api
Message-ID: <99b1e8d2d2e94bda68c53178aaf6f04d@mail.gmail.com>

AFAIK the % let's Freeswitch know it's a local domain and doesn't need to do
any DNS lookups, etc. While it was never really necessary to use % it made a
lot sense for various reasons.


--
Kristian Kielhofner
http://blog.krisk.org

----- Original Message -----
From: freeswitch-users-bounces at lists.freeswitch.org
<freeswitch-users-bounces at lists.freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
<freeswitch-users at lists.freeswitch.org>
Sent: Fri May 21 12:07:17 2010
Subject: Re: [Freeswitch-users] Quick question on Originate through api

not necessary ....

----- Original Message -----
From: "Peter Olsson" <peter.olsson at visionutveckling.se>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Friday, May 21, 2010 11:52 AM
Subject: Re: [Freeswitch-users] Quick question on Originate through api


For sofia/internal/ you should use a % as a separator I think, instead of
@-sign.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org
[freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Phillip Jones
[pjintheusa at gmail.com]
Skickat: den 21 maj 2010 17:41
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] Quick question on Originate through api

Hi there,

In my mind - where 6775557767 is a registered user, the following two are
the same:

apiResult = fsApi.Execute("originate",
string.Format("{{ignore_early_media=false,absolute_codec_string='PCMU',origination_caller_id_number=5556667776}}user/6775557767
'

apiResult = fsApi.Execute("originate",
string.Format("{{ignore_early_media=false,absolute_codec_string='PCMU',origination_caller_id_number=5556667776}}sofia/internal/6775557767 at 121.763.76.76
'

i.e. user/ and sofia/internal are the same thing.

However on my system the user/ works great but sofia/internal loops call
back through in the public context.

Can someone tell me the difference between these options?

What am I missing?

Thanks


Phil



!DSPAM:4bf6abb932934294689129!

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


_______________________________________________
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http://www.freeswitch.org


From stevendt at primrosebank.net  Fri May 21 09:52:42 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Fri, 21 May 2010 17:52:42 +0100
Subject: [Freeswitch-users] Mod_skinny: Call for testing!
References: <AANLkTik8hHRtOci8qTXmZVHOWvVxRg-OPm5zyRAMrWEe@mail.gmail.com><F32D0E59CC994113BC88C7902AFC87BF@bp1.ad.bp.com>,
	<AANLkTikJ1biYWWY-M4qYMWgsYfGelxvFu91EVnBqDlcq@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C557DE57F78@cooper>
Message-ID: <3A8B4A19CB3C4270B20D9980B9E02F45@bp1.ad.bp.com>

Hi,

thanks Peter.

I did not realise that the Windows build file was so new, OK, I guess that 
I'll just have to wait for the SVN to catch up. I don't really want to 
update to Git at the moment,

regards
Dave

----- Original Message ----- 
From: "Peter Olsson" <peter.olsson at visionutveckling.se>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Friday, May 21, 2010 4:16 PM
Subject: Re: [Freeswitch-users] Mod_skinny: Call for testing!


I tried building today, and it builds on Windows on latest git - just needs 
some testing now :)

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org 
[freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Mathieu Parent 
[math.parent at gmail.com]
Skickat: den 21 maj 2010 16:24
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Mod_skinny: Call for testing!

On Fri, May 21, 2010 at 3:46 PM, Dave Stevenson
<stevendt at primrosebank.net> wrote:
> Mathieu,
>
> just updated to 17539 (SVN) and skinny is not included in the Windows 
> Build
> Solution to VS2008
>
> Does it need to be compiled & linked outside of the VS Build ?

SVN is out of sync, git has Windows support since yesterday:

commit bfb6a8b1f5f835008d1e0fbd90d2e120814f341a
Author: Mathieu Parent <math.parent at gmail.com>
Date:   Thu May 20 23:19:10 2010 +0200

    Skinny: Initial Windows support

    Thanks to Peter Olsson

    Closes: MODSKINNY-3

I have not been able to test it, you can comment MODSKINNY-3 if you
still have problem after the patch applied.

Mathieu Parent

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

!DSPAM:4bf6994c32931601618424!


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



From testa at voicetechnology.com.br  Fri May 21 10:06:30 2010
From: testa at voicetechnology.com.br (Fernando Testa)
Date: Fri, 21 May 2010 14:06:30 -0300
Subject: [Freeswitch-users] Mod_skinny: Call for testing!
In-Reply-To: <AANLkTik8hHRtOci8qTXmZVHOWvVxRg-OPm5zyRAMrWEe@mail.gmail.com>
References: <AANLkTik8hHRtOci8qTXmZVHOWvVxRg-OPm5zyRAMrWEe@mail.gmail.com>
Message-ID: <AANLkTim5i2oRObH9Q_-ADCNBbIvR__LVb7qOOX6jhzFF@mail.gmail.com>

Mathieu, Thank you for this. Can I use FS in an extension position?
I would like to put FS as voicemail system, behind a cisco pbx.

On Thu, May 20, 2010 at 9:54 PM, Mathieu Parent <math.parent at gmail.com>wrote:

> Hello,
>
> Skinny Call Control Protocol (SCCP) is a VoIP protocol used on several
> phones including Cisco IP Phones and Nokia Call Connect 2.0. Contrary
> to SIP, the phone is controlled by the server. This allow powerfull
> and funny things (like controlling leds, display, ring type, ...) ;
> this also allow to use some phones with crappy SIP images, like (guess
> what?) Cisco ones.
>
> Since February, mod_skinny is in freeswitch trunk. Many things
> happened since. Today the module has reached several milestone (it
> compiles on windows, call info is shown early, state handling changed
> to a more FreeSWITCH manner, ...).
>
> Most of basic features and more are implemented (from Wiki):
>    * Incoming and outgoing calls
>    * Shared lines and multiple calls per lines
>    * CallInfo set as early as possible
>    * New call made from: OffHook, Softkeys (Redial, NewCall),
> Stimulus (LastNumberRedial, VoiceMail, SpeedDial)
>    * Answering calls via: OffHook, Softkeys (Answer)
>    * Holding/Unholding calls via Softkeys (Hold, Resume) or Stimulus
> (Hold), with MOH (music on hold)
>    * Transfer via Softkeys (Transfer) or Stimulus (Transfer)
>    * Ending call via: OnHook, Softkeys (EndCall)
>    * Misc: SoftKeys (Backspace)
>    * Handle firmware version request (VersionReqMessage) per device
> type or per device
>    * Patterns for dialed numbers to process (patterns-dialplan and
> patterns-context)
>
> Some important missing ones:
> * Message Waiting indicator (MWI)
> * Bypass Media (help needed)
> * Codec Negotiation (help needed)
> * Forward (transfer already implemented)
> * Complete list:
> <http://wiki.freeswitch.org/wiki/Skinny_Call_Control_Protocol#TODO>
>
> How can you help:
> * test if it compiles on your platform
> * test if it works with your SCCP phone (only 7960 and 7961 have been
> tested yet)
> * if you don't have an SCCP phone, test using "perl
> src/mod/endpoints/mod_skinny/test-skinny.pl" (you need a recent perl,
> aka 5.10.1, 5.10.0 has buggy thread implementation, at least on Debian
> Lenny).
> * write docs about configuration
> * propose patches (some are easy like API Commands or events ; some
> are harder ;)
> * review the code (quality, security, ...)
> * If you own a Cisco Communication Manager and you know a feature that
> is not in mod_skinny, make network captures!
>
> Some links:
> * Instructions are at: http://wiki.freeswitch.org/wiki/Mod_skinny
> * Bugs should go here: http://jira.freeswitch.org/browse/MODSKINNY
> * Development stuff is at:
> http://wiki.freeswitch.org/wiki/Skinny_Call_Control_Protocol
> * Suggested features should go here:
> http://wiki.freeswitch.org/wiki/Skinny_Call_Control_Protocol#TODO (or
> bounty or JIRA)
>
> Any help is appreciated.
>
> happy testing/hacking ;)
>
> Mathieu Parent
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Fernando Gregianin Testa
Voice Technology Ltda
+55 11 35882166
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From pjintheusa at gmail.com  Fri May 21 10:08:41 2010
From: pjintheusa at gmail.com (Phillip Jones)
Date: Fri, 21 May 2010 13:08:41 -0400
Subject: [Freeswitch-users] Quick question on Originate through api
In-Reply-To: <99b1e8d2d2e94bda68c53178aaf6f04d@mail.gmail.com>
References: <99b1e8d2d2e94bda68c53178aaf6f04d@mail.gmail.com>
Message-ID: <AANLkTimThs22VWq8tGOSws2PjfnqrgzM5GRV6fhuUYVs@mail.gmail.com>

Hmm...

That is it. % vs @ is the answer.

Use @ the call loops back through and brings my world and FreeSWITCH
crashing down.

Use % and all is well in the world and the call is routed.

Obviously I need to understand this better.

On Fri, May 21, 2010 at 12:40 PM, Kristian Kielhofner <kris at kriskinc.com>wrote:

> AFAIK the % let's Freeswitch know it's a local domain and doesn't need to
> do
> any DNS lookups, etc. While it was never really necessary to use % it made
> a
> lot sense for various reasons.
>
>
> --
> Kristian Kielhofner
> http://blog.krisk.org
>
> ----- Original Message -----
> From: freeswitch-users-bounces at lists.freeswitch.org
> <freeswitch-users-bounces at lists.freeswitch.org>
> To: freeswitch-users at lists.freeswitch.org
> <freeswitch-users at lists.freeswitch.org>
> Sent: Fri May 21 12:07:17 2010
> Subject: Re: [Freeswitch-users] Quick question on Originate through api
>
> not necessary ....
>
> ----- Original Message -----
> From: "Peter Olsson" <peter.olsson at visionutveckling.se>
> To: <freeswitch-users at lists.freeswitch.org>
> Sent: Friday, May 21, 2010 11:52 AM
> Subject: Re: [Freeswitch-users] Quick question on Originate through api
>
>
> For sofia/internal/ you should use a % as a separator I think, instead of
> @-sign.
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
> [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Phillip Jones
> [pjintheusa at gmail.com]
> Skickat: den 21 maj 2010 17:41
> Till: freeswitch-users at lists.freeswitch.org
> ?mne: [Freeswitch-users] Quick question on Originate through api
>
> Hi there,
>
> In my mind - where 6775557767 is a registered user, the following two are
> the same:
>
> apiResult = fsApi.Execute("originate",
>
> string.Format("{{ignore_early_media=false,absolute_codec_string='PCMU',origination_caller_id_number=5556667776}}user/6775557767
> '
>
> apiResult = fsApi.Execute("originate",
>
> string.Format("{{ignore_early_media=false,absolute_codec_string='PCMU',origination_caller_id_number=5556667776}}sofia/internal/6775557767 at 121.763.76.76
> '
>
> i.e. user/ and sofia/internal are the same thing.
>
> However on my system the user/ works great but sofia/internal loops call
> back through in the public context.
>
> Can someone tell me the difference between these options?
>
> What am I missing?
>
> Thanks
>
>
> Phil
>
>
>
> !DSPAM:4bf6abb932934294689129!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From math.parent at gmail.com  Fri May 21 10:19:34 2010
From: math.parent at gmail.com (Mathieu Parent)
Date: Fri, 21 May 2010 19:19:34 +0200
Subject: [Freeswitch-users] Mod_skinny: Call for testing!
In-Reply-To: <AANLkTim5i2oRObH9Q_-ADCNBbIvR__LVb7qOOX6jhzFF@mail.gmail.com>
References: <AANLkTik8hHRtOci8qTXmZVHOWvVxRg-OPm5zyRAMrWEe@mail.gmail.com>
	<AANLkTim5i2oRObH9Q_-ADCNBbIvR__LVb7qOOX6jhzFF@mail.gmail.com>
Message-ID: <AANLkTil7W6tgqfh5Pd5DMynbY70C2_IYag0uQ81COhRH@mail.gmail.com>

On Fri, May 21, 2010 at 7:06 PM, Fernando Testa
<testa at voicetechnology.com.br> wrote:
> Mathieu, Thank you for this. Can I use FS in an extension position?
> I would like to put FS as voicemail system, behind a cisco pbx.

For this, you don't need mod_skinny. Prefer a SIP trunk between FS and
CCM. Instructions for Asterisk are here:
<http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Integration>,
it can be adapted for FreeSWITCH; See also
http://wiki.freeswitch.org/wiki/Cisco_Call_Manager and list archives.

Mod_skinny is a replacement to CCM and not a replacement to Cisco IP
Phones (i.e mod_skinny implements server part but not client part).

Mathieu Parent


From peter.olsson at visionutveckling.se  Fri May 21 10:20:06 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 21 May 2010 19:20:06 +0200
Subject: [Freeswitch-users] Mod_skinny: Call for testing!
In-Reply-To: <AANLkTim5i2oRObH9Q_-ADCNBbIvR__LVb7qOOX6jhzFF@mail.gmail.com>
References: <AANLkTik8hHRtOci8qTXmZVHOWvVxRg-OPm5zyRAMrWEe@mail.gmail.com>,
	<AANLkTim5i2oRObH9Q_-ADCNBbIvR__LVb7qOOX6jhzFF@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C557DE57F7B@cooper>

I think you're better off using SIP trunking in that case. We're running FS behind Cisco for this scenario, using SIP, and it works perfectly.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Fernando Testa [testa at voicetechnology.com.br]
Skickat: den 21 maj 2010 19:06
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Mod_skinny: Call for testing!

Mathieu, Thank you for this. Can I use FS in an extension position?
I would like to put FS as voicemail system, behind a cisco pbx.

On Thu, May 20, 2010 at 9:54 PM, Mathieu Parent <math.parent at gmail.com<mailto:math.parent at gmail.com>> wrote:
Hello,

Skinny Call Control Protocol (SCCP) is a VoIP protocol used on several
phones including Cisco IP Phones and Nokia Call Connect 2.0. Contrary
to SIP, the phone is controlled by the server. This allow powerfull
and funny things (like controlling leds, display, ring type, ...) ;
this also allow to use some phones with crappy SIP images, like (guess
what?) Cisco ones.

Since February, mod_skinny is in freeswitch trunk. Many things
happened since. Today the module has reached several milestone (it
compiles on windows, call info is shown early, state handling changed
to a more FreeSWITCH manner, ...).

Most of basic features and more are implemented (from Wiki):
   * Incoming and outgoing calls
   * Shared lines and multiple calls per lines
   * CallInfo set as early as possible
   * New call made from: OffHook, Softkeys (Redial, NewCall),
Stimulus (LastNumberRedial, VoiceMail, SpeedDial)
   * Answering calls via: OffHook, Softkeys (Answer)
   * Holding/Unholding calls via Softkeys (Hold, Resume) or Stimulus
(Hold), with MOH (music on hold)
   * Transfer via Softkeys (Transfer) or Stimulus (Transfer)
   * Ending call via: OnHook, Softkeys (EndCall)
   * Misc: SoftKeys (Backspace)
   * Handle firmware version request (VersionReqMessage) per device
type or per device
   * Patterns for dialed numbers to process (patterns-dialplan and
patterns-context)

Some important missing ones:
* Message Waiting indicator (MWI)
* Bypass Media (help needed)
* Codec Negotiation (help needed)
* Forward (transfer already implemented)
* Complete list:
<http://wiki.freeswitch.org/wiki/Skinny_Call_Control_Protocol#TODO>

How can you help:
* test if it compiles on your platform
* test if it works with your SCCP phone (only 7960 and 7961 have been
tested yet)
* if you don't have an SCCP phone, test using "perl
src/mod/endpoints/mod_skinny/test-skinny.pl<http://test-skinny.pl>" (you need a recent perl,
aka 5.10.1, 5.10.0 has buggy thread implementation, at least on Debian
Lenny).
* write docs about configuration
* propose patches (some are easy like API Commands or events ; some
are harder ;)
* review the code (quality, security, ...)
* If you own a Cisco Communication Manager and you know a feature that
is not in mod_skinny, make network captures!

Some links:
* Instructions are at: http://wiki.freeswitch.org/wiki/Mod_skinny
* Bugs should go here: http://jira.freeswitch.org/browse/MODSKINNY
* Development stuff is at:
http://wiki.freeswitch.org/wiki/Skinny_Call_Control_Protocol
* Suggested features should go here:
http://wiki.freeswitch.org/wiki/Skinny_Call_Control_Protocol#TODO (or
bounty or JIRA)

Any help is appreciated.

happy testing/hacking ;)

Mathieu Parent

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



--
Fernando Gregianin Testa
Voice Technology Ltda
+55 11 35882166

!DSPAM:4bf6bf7d32931196818963!


From david.ponzone at gmail.com  Fri May 21 10:42:35 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Fri, 21 May 2010 19:42:35 +0200
Subject: [Freeswitch-users] Quick question on Originate through api
In-Reply-To: <AANLkTimThs22VWq8tGOSws2PjfnqrgzM5GRV6fhuUYVs@mail.gmail.com>
References: <99b1e8d2d2e94bda68c53178aaf6f04d@mail.gmail.com>
	<AANLkTimThs22VWq8tGOSws2PjfnqrgzM5GRV6fhuUYVs@mail.gmail.com>
Message-ID: <A94E5492-27FD-4C92-B695-AA9F7ED29F58@gmail.com>

May I use the opportunity to ask something more or less related to  
that ?

I use sofia/profile/user%domain^XXXXX to modify the SIP TO in the  
INVITE sent to a registered user.
I was expecting the syntax: user/<user> ^XXXXXX to work also, but it  
doesn't.

Is that normal ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 21/05/2010 ? 19:08, Phillip Jones a ?crit :

> Hmm...
>
> That is it. % vs @ is the answer.
>
> Use @ the call loops back through and brings my world and FreeSWITCH  
> crashing down.
>
> Use % and all is well in the world and the call is routed.
>
> Obviously I need to understand this better.
>
> On Fri, May 21, 2010 at 12:40 PM, Kristian Kielhofner <kris at kriskinc.com 
> > wrote:
> AFAIK the % let's Freeswitch know it's a local domain and doesn't  
> need to do
> any DNS lookups, etc. While it was never really necessary to use %  
> it made a
> lot sense for various reasons.
>
>
> --
> Kristian Kielhofner
> http://blog.krisk.org
>
> ----- Original Message -----
> From: freeswitch-users-bounces at lists.freeswitch.org
> <freeswitch-users-bounces at lists.freeswitch.org>
> To: freeswitch-users at lists.freeswitch.org
> <freeswitch-users at lists.freeswitch.org>
> Sent: Fri May 21 12:07:17 2010
> Subject: Re: [Freeswitch-users] Quick question on Originate through  
> api
>
> not necessary ....
>
> ----- Original Message -----
> From: "Peter Olsson" <peter.olsson at visionutveckling.se>
> To: <freeswitch-users at lists.freeswitch.org>
> Sent: Friday, May 21, 2010 11:52 AM
> Subject: Re: [Freeswitch-users] Quick question on Originate through  
> api
>
>
> For sofia/internal/ you should use a % as a separator I think,  
> instead of
> @-sign.
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
> [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Phillip Jones
> [pjintheusa at gmail.com]
> Skickat: den 21 maj 2010 17:41
> Till: freeswitch-users at lists.freeswitch.org
> ?mne: [Freeswitch-users] Quick question on Originate through api
>
> Hi there,
>
> In my mind - where 6775557767 is a registered user, the following  
> two are
> the same:
>
> apiResult = fsApi.Execute("originate",
> string 
> .Format 
> ("{{ignore_early_media 
> = 
> false 
> ,absolute_codec_string 
> ='PCMU',origination_caller_id_number=5556667776}}user/6775557767
> '
>
> apiResult = fsApi.Execute("originate",
> string 
> .Format("{{ignore_early_media=false,absolute_codec_string='PCMU',origination_caller_id_number=5556667776}}sofia/internal/6775557767 at 121.763.76.76
> '
>
> i.e. user/ and sofia/internal are the same thing.
>
> However on my system the user/ works great but sofia/internal loops  
> call
> back through in the public context.
>
> Can someone tell me the difference between these options?
>
> What am I missing?
>
> Thanks
>
>
> Phil
>
>
>
> !DSPAM:4bf6abb932934294689129!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From samwise+fsu at bagshot-row.org  Fri May 21 09:03:22 2010
From: samwise+fsu at bagshot-row.org (samwise)
Date: Fri, 21 May 2010 17:03:22 +0100
Subject: [Freeswitch-users] ISDN to SIP via Freeswitch - 408 Timeout and
	Terrible line noise
Message-ID: <AANLkTiliTh0KWxn_ylskkVU8kwfBz4e_ji_m4v6xhfJ7@mail.gmail.com>

Hi,

I've just joined a team which had an Asterisk 1.6.0.1 desktop machine
with a Digium TE122 card running CentOS 4.7 to bridge calls from, as
far as I can tell, an E1 PRI line (with 10 DDI numbers) and software
SIP UAs.  Unfortunately, the machine died recently and will no longer
boot from hard disk.

So I've taken the card out and placed it in a HP Compaq ProLiant ML350
G3 Server with 1 GB RAM running CentOS 5.4 (Final).  I've set it up to
run the Freeswitch 1.0.6 release with the OpenZAP module enabled and
pre-requisites dahdi-linux-complete-2.3.0+2.3.0 and libpri-1.4.10.2.
All these components were built from source packages.

I installed the default configuration files and configured everything
as far as I could by following the instructions on the Freeswitch
wiki.  I've managed to get as far as successfully being able to dial
one of the DDI numbers and getting the call to be delivered to an
X-Lite SIP UA I have registered with Freeswitch.

Unfortunately, I have two issues:

1) I have configured a SIP user account (Greg) but when I try to
register X-Lite, it consistently returns a Registration error: 408 -
Request Timeout.  However, if I leave X-Lite alone, most of the time
it will eventually (after quite some time) re-register with the server
of it's own accord, allowing me to try a test call from the PSTN.

2) Whilst the SIP UA can then hear the call fine, there is a *lot* of
noise on the PSTN end of the call - so much so that it's basically
unusable, even though I can hear some things from the SIP end.

Some info on the kernel/CPU included belom and some possibly useful
logs / config files available here:

openzap.conf: http://pastebin.freeswitch.org/13019
openzap.conf.xml: http://pastebin.freeswitch.org/13020
Greg.xml: http://pastebin.freeswitch.org/13021
61522x.xml: http://pastebin.freeswitch.org/13022
freeswitch.log: http://pastebin.freeswitch.org/13023

My question is where to start looking to diagnose these problems?  I
can provide whatever logs may be appropriate though I can't see any
major problems in the freeswitch.log currently.

With regards to the line noise issue, I read in the OpenZAP FAQ that
unlike Asterisk, Freeswitch does not include a software echo
canceller.  Could that be related to the line noise?  Should I attempt
to install OSLEC to improve the quality?

Thanks for any help - I'm fairly new to the world of telephony
tecnologies like ISDN,

Sam.

# uname -a
Linux bob 2.6.18-164.el5 #1 SMP Thu Sep 3 03:33:56 EDT 2009 i686 i686
i386 GNU/Linux
# cat /proc/cpuinfo
processor       : 0
vendor_id       : GenuineIntel
cpu family      : 6
model           : 8
model name      : Pentium III (Coppermine)
stepping        : 6
cpu MHz         : 864.011
cache size      : 256 KB
fdiv_bug        : no
hlt_bug         : no
f00f_bug        : no
coma_bug        : no
fpu             : yes
fpu_exception   : yes
cpuid level     : 2
wp              : yes
flags           : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca
cmov pat pse36 mmx fxsr sse up
bogomips        : 1728.02


From jan.berger at video24.no  Fri May 21 11:57:55 2010
From: jan.berger at video24.no (Jan Berger)
Date: Fri, 21 May 2010 20:57:55 +0200
Subject: [Freeswitch-users] mobisents
Message-ID: <D0A05C002FC345968822B6CCE907B618@dell9400>

Hi,

 

Is anyone working on a FreeSWITCH RA for mobicents?

 

I just realised that mobicents already have sccp, tcap, inap, camel, map + a
few other things. They already have a Asterisk RA.

 

Jan

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From anthony.minessale at gmail.com  Fri May 21 12:55:17 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 21 May 2010 14:55:17 -0500
Subject: [Freeswitch-users] Quick question on Originate through api
In-Reply-To: <A94E5492-27FD-4C92-B695-AA9F7ED29F58@gmail.com>
References: <99b1e8d2d2e94bda68c53178aaf6f04d@mail.gmail.com>
	<AANLkTimThs22VWq8tGOSws2PjfnqrgzM5GRV6fhuUYVs@mail.gmail.com>
	<A94E5492-27FD-4C92-B695-AA9F7ED29F58@gmail.com>
Message-ID: <AANLkTin2FryNdgKuNOjpG5OWpAsp93fvolGcbFliBy4v@mail.gmail.com>

the user endpoint has no idea what ^XXXXX means

try the newer version of that ^ thing which is:

{sip_invite_to_uri=XXXXX}user/<user>

On Fri, May 21, 2010 at 12:42 PM, David Ponzone <david.ponzone at gmail.com>wrote:

> May I use the opportunity to ask something more or less related to that ?
>
> I use sofia/profile/user%domain^XXXXX to modify the SIP TO in the INVITE
> sent to a registered user.
> I was expecting the syntax: user/<user> ^XXXXXX to work also, but it
> doesn't.
>
> Is that normal ?
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 21/05/2010 ? 19:08, Phillip Jones a ?crit :
>
> Hmm...
>
> That is it. % vs @ is the answer.
>
> Use @ the call loops back through and brings my world and FreeSWITCH
> crashing down.
>
> Use % and all is well in the world and the call is routed.
>
> Obviously I need to understand this better.
>
> On Fri, May 21, 2010 at 12:40 PM, Kristian Kielhofner <kris at kriskinc.com>wrote:
>
>> AFAIK the % let's Freeswitch know it's a local domain and doesn't need to
>> do
>> any DNS lookups, etc. While it was never really necessary to use % it made
>> a
>> lot sense for various reasons.
>>
>>
>> --
>> Kristian Kielhofner
>> http://blog.krisk.org
>>
>> ----- Original Message -----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> <freeswitch-users-bounces at lists.freeswitch.org>
>> To: freeswitch-users at lists.freeswitch.org
>> <freeswitch-users at lists.freeswitch.org>
>> Sent: Fri May 21 12:07:17 2010
>> Subject: Re: [Freeswitch-users] Quick question on Originate through api
>>
>> not necessary ....
>>
>> ----- Original Message -----
>> From: "Peter Olsson" <peter.olsson at visionutveckling.se>
>> To: <freeswitch-users at lists.freeswitch.org>
>> Sent: Friday, May 21, 2010 11:52 AM
>> Subject: Re: [Freeswitch-users] Quick question on Originate through api
>>
>>
>> For sofia/internal/ you should use a % as a separator I think, instead of
>> @-sign.
>>
>> /Peter
>> ________________________________________
>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
>> [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Phillip Jones
>> [pjintheusa at gmail.com]
>> Skickat: den 21 maj 2010 17:41
>> Till: freeswitch-users at lists.freeswitch.org
>> ?mne: [Freeswitch-users] Quick question on Originate through api
>>
>> Hi there,
>>
>> In my mind - where 6775557767 is a registered user, the following two are
>> the same:
>>
>> apiResult = fsApi.Execute("originate",
>>
>> string.Format("{{ignore_early_media=false,absolute_codec_string='PCMU',origination_caller_id_number=5556667776}}user/6775557767
>> '
>>
>> apiResult = fsApi.Execute("originate",
>> string.Format("{{ignore_early_media=false,absolute_codec_string='PCMU',
>> origination_caller_id_number=5556667776}}sofia/internal/6775557767 at 121.763.76.76<origination_caller_id_number=5556667776%7D%7Dsofia/internal/6775557767 at 121.763.76.76>
>> '
>>
>> i.e. user/ and sofia/internal are the same thing.
>>
>> However on my system the user/ works great but sofia/internal loops call
>> back through in the public context.
>>
>> Can someone tell me the difference between these options?
>>
>> What am I missing?
>>
>> Thanks
>>
>>
>> Phil
>>
>>
>>
>> !DSPAM:4bf6abb932934294689129!
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
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
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From msc at freeswitch.org  Fri May 21 13:20:22 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 21 May 2010 13:20:22 -0700
Subject: [Freeswitch-users] mod_dptools/redirect and 300 Multiple Choices
In-Reply-To: <AANLkTinYtV2CL1H_hME51kP4OpJLaYeuQK0jUEnrimAz@mail.gmail.com>
References: <AANLkTinYtV2CL1H_hME51kP4OpJLaYeuQK0jUEnrimAz@mail.gmail.com>
Message-ID: <AANLkTimD5crX5mzwO-fwyvKgKfKP3EozxTrwyVivMtgg@mail.gmail.com>

On Thu, May 20, 2010 at 11:46 PM, Carlos Romero <romerocarlos24 at gmail.com>wrote:

> what do you mean that you don't have access to it?
>
> you have the code
>

The SIP stack is supplied by Nokia. Open source or not, it is bad form to
muck with someone else's library without due cause. Like all robust SIP
stacks, Sofia is huge, deep, complex, and otherwise uninviting of those who
would waltz in and do whatever they wish. If you feel that it is even within
the realm of possibility then I recommend that you find the location in the
Sofia library where this takes place, make the appropriate changes, and
thoroughly test for regressions. Submit the patch to the Sofia devs who will
then evaluate it and see if it is worthy of inclusion.

Or more simply put: "having the code" and "having access" are COMPLETELY
different concepts.
-MC


>
> On Thu, May 20, 2010 at 3:23 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
> > not without intervention from the sofia devs,
> > Sofia sip does that deep down and we don't have access to it.
> > even when we supply it as a one big string it renders it out depending on
> > those settings.
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Fri May 21 13:38:13 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 21 May 2010 15:38:13 -0500
Subject: [Freeswitch-users] mod_dptools/redirect and 300 Multiple Choices
In-Reply-To: <AANLkTimD5crX5mzwO-fwyvKgKfKP3EozxTrwyVivMtgg@mail.gmail.com>
References: <AANLkTinYtV2CL1H_hME51kP4OpJLaYeuQK0jUEnrimAz@mail.gmail.com>
	<AANLkTimD5crX5mzwO-fwyvKgKfKP3EozxTrwyVivMtgg@mail.gmail.com>
Message-ID: <AANLkTinuRnLgvYNHWfYWP7NT7ANi2H5e_5kiWvNkwg5E@mail.gmail.com>

What he said....

P. S.
In this context, "access to it" means weather or not the intended API
provides the desired functionality.



On Fri, May 21, 2010 at 3:20 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Thu, May 20, 2010 at 11:46 PM, Carlos Romero <romerocarlos24 at gmail.com>wrote:
>
>> what do you mean that you don't have access to it?
>>
>> you have the code
>>
>
> The SIP stack is supplied by Nokia. Open source or not, it is bad form to
> muck with someone else's library without due cause. Like all robust SIP
> stacks, Sofia is huge, deep, complex, and otherwise uninviting of those who
> would waltz in and do whatever they wish. If you feel that it is even within
> the realm of possibility then I recommend that you find the location in the
> Sofia library where this takes place, make the appropriate changes, and
> thoroughly test for regressions. Submit the patch to the Sofia devs who will
> then evaluate it and see if it is worthy of inclusion.
>
> Or more simply put: "having the code" and "having access" are COMPLETELY
> different concepts.
> -MC
>
>
>>
>> On Thu, May 20, 2010 at 3:23 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>> > not without intervention from the sofia devs,
>> > Sofia sip does that deep down and we don't have access to it.
>> > even when we supply it as a one big string it renders it out depending
>> on
>> > those settings.
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
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
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From codeghar at gmail.com  Fri May 21 18:17:15 2010
From: codeghar at gmail.com (Code Ghar)
Date: Fri, 21 May 2010 20:17:15 -0500
Subject: [Freeswitch-users] CDRs for Gateways
Message-ID: <AANLkTimSu_Kglb4y5vkFAi0MlarfkVc45LuGwW9erKRt@mail.gmail.com>

Hi David

I looked at what you suggested. Before your suggestion, in
/usr/local/freeswitch/conf/autoload_configs/cdr_csv.conf file, I had the
following lines


<!-- Only log in Master.csv -->
<!--<param name="master-file-only" value="true"/>-->

After your suggestion, I changed it to

<!-- Only log in Master.csv -->
<!--<param name="master-file-only" value="true"/>-->

and then did a reloadxml. Now all CDRs were created in
/usr/local/freeswitch/log/cdr-csv/Master.csv and CDRs for inbound calls from
unregistered sources (i.e. external gateways) were also created in this
file. Thanks for your help.



Code,

I think you have a major issue or configuration error with your FS
setup.
FS, as any B2BUA, generates a CDR for any inbound leg, either from a
registered user or from a non-registered gateway.
It doesn't care where it comes from, it just does it.
I can barely imagine a softswitch that would not write the CDR
matching a call that was routed through it.

The CDR for the outbound leg is optional.

Can you confirm you are looking in Master.csv ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
? l'intention exclusive de ses destinataires. Toute utilisation ou
diffusion non autoris?e est interdite. Tout message ?lectronique est
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes
pas destinataire de ce message, merci de le d?truire imm?diatement et
d'avertir l'exp?diteur.




Le 21/05/2010 ? 05:25, Code Ghar a ?crit :

> Using mod_cdr_csv, it is easy to create CDRs for registered users.
> However, how can we get CDRs for external gateways? For example,
> FreeSWITCH is configured to receive calls from one gateway and send
> them out through another. In this scenario, when neither inbound nor
> outbound gateway registers itself to FreeSWITCH, how can we generate
> and store CDRs for inbound gateway?
>
> Similarly, let's say a SIP provider sends a call meant for a
> registered user. For example, if a cell phone calls 6175550000, it
> is routed through the telephone network to the SIP provider. This
> provider routes the call to FreeSWITCH, which looks up in its dial
> plan that this number should be routed to extension (or registered
> user) 1000. There are no CDRs generated for either the SIP
> provider's gateway or the registered user in this case.
>
> In both these scenarios, if a call is inbound to FreeSWITCH but has
> not been generated by a registered user, mod_cdr_csv does not create
> CDRs. I have even tried to set the following parameter in
> cdr_csv.conf.xml      but it didn't help.
>
> <param name="legs" value="ab"/>
>
> My question is this: can mod_cdr_csv generate CDRs for inbound calls
> from sources other than registered users? If it can't, how can this
> be achieved?
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From peter.olsson at visionutveckling.se  Fri May 21 23:41:34 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sat, 22 May 2010 08:41:34 +0200
Subject: [Freeswitch-users] Conference command "relate", nospeak/nohear
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C557DE57F81@cooper>

Hello everyone,

I have a small question about the mod_conference relate command. What's the expected behaviour of "nohear"? Is it supposed to work as nospeak, but in the other direction?

When running "conference xx relate member_x member_y nospeak", the audio from member_x to member_y is muted, but when using nohear I didn't notice any difference at all (I could still hear audio everywhere). Is it supposed to work the opposite way, so when using nospeak|nohear as parameter, the audio from x to y is muted, and also the other way around?

/Peter


From david.ponzone at gmail.com  Sat May 22 00:33:49 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Sat, 22 May 2010 09:33:49 +0200
Subject: [Freeswitch-users] CDRs for Gateways
In-Reply-To: <AANLkTimSu_Kglb4y5vkFAi0MlarfkVc45LuGwW9erKRt@mail.gmail.com>
References: <AANLkTimSu_Kglb4y5vkFAi0MlarfkVc45LuGwW9erKRt@mail.gmail.com>
Message-ID: <63B4B0C5-B499-4002-A967-567417AE616D@gmail.com>

Well, according to your mail, you did not change anything :)
I guess it is a typo.

Can you confirm what you changed ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 22/05/2010 ? 03:17, Code Ghar a ?crit :

> Hi David
>
> I looked at what you suggested. Before your suggestion, in /usr/ 
> local/freeswitch/conf/autoload_configs/cdr_csv.conf file, I had the  
> following lines
>
>
> <!-- Only log in Master.csv -->
> <!--<param name="master-file-only" value="true"/>-->
>
> After your suggestion, I changed it to
>
> <!-- Only log in Master.csv -->
> <!--<param name="master-file-only" value="true"/>-->
>
> and then did a reloadxml. Now all CDRs were created in /usr/local/ 
> freeswitch/log/cdr-csv/Master.csv and CDRs for inbound calls from  
> unregistered sources (i.e. external gateways) were also created in  
> this file. Thanks for your help.
>
>
>
> Code,
>
> I think you have a major issue or configuration error with your FS
> setup.
> FS, as any B2BUA, generates a CDR for any inbound leg, either from a
> registered user or from a non-registered gateway.
> It doesn't care where it comes from, it just does it.
> I can barely imagine a softswitch that would not write the CDR
> matching a call that was routed through it.
>
> The CDR for the outbound leg is optional.
>
> Can you confirm you are looking in Master.csv ?
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
> ? l'intention exclusive de ses destinataires. Toute utilisation ou
> diffusion non autoris?e est interdite. Tout message ?lectronique est
> susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre
> de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes
> pas destinataire de ce message, merci de le d?truire imm?diatement et
> d'avertir l'exp?diteur.
>
>
>
>
> Le 21/05/2010 ? 05:25, Code Ghar a ?crit :
>
> > Using mod_cdr_csv, it is easy to create CDRs for registered users.
> > However, how can we get CDRs for external gateways? For example,
> > FreeSWITCH is configured to receive calls from one gateway and send
> > them out through another. In this scenario, when neither inbound nor
> > outbound gateway registers itself to FreeSWITCH, how can we generate
> > and store CDRs for inbound gateway?
> >
> > Similarly, let's say a SIP provider sends a call meant for a
> > registered user. For example, if a cell phone calls 6175550000, it
> > is routed through the telephone network to the SIP provider. This
> > provider routes the call to FreeSWITCH, which looks up in its dial
> > plan that this number should be routed to extension (or registered
> > user) 1000. There are no CDRs generated for either the SIP
> > provider's gateway or the registered user in this case.
> >
> > In both these scenarios, if a call is inbound to FreeSWITCH but has
> > not been generated by a registered user, mod_cdr_csv does not create
> > CDRs. I have even tried to set the following parameter in
> > cdr_csv.conf.xml      but it didn't help.
> >
> > <param name="legs" value="ab"/>
> >
> > My question is this: can mod_cdr_csv generate CDRs for inbound calls
> > from sources other than registered users? If it can't, how can this
> > be achieved?
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
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>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From david.ponzone at gmail.com  Sat May 22 01:07:16 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Sat, 22 May 2010 10:07:16 +0200
Subject: [Freeswitch-users] Quick question on Originate through api
In-Reply-To: <AANLkTin2FryNdgKuNOjpG5OWpAsp93fvolGcbFliBy4v@mail.gmail.com>
References: <99b1e8d2d2e94bda68c53178aaf6f04d@mail.gmail.com>
	<AANLkTimThs22VWq8tGOSws2PjfnqrgzM5GRV6fhuUYVs@mail.gmail.com>
	<A94E5492-27FD-4C92-B695-AA9F7ED29F58@gmail.com>
	<AANLkTin2FryNdgKuNOjpG5OWpAsp93fvolGcbFliBy4v@mail.gmail.com>
Message-ID: <51B2AC80-DCED-4FD7-B219-096C8BF3C5F0@gmail.com>

Thanks Anthony!

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 21/05/2010 ? 21:55, Anthony Minessale a ?crit :

> the user endpoint has no idea what ^XXXXX means
>
> try the newer version of that ^ thing which is:
>
> {sip_invite_to_uri=XXXXX}user/<user>
>
> On Fri, May 21, 2010 at 12:42 PM, David Ponzone <david.ponzone at gmail.com 
> > wrote:
> May I use the opportunity to ask something more or less related to  
> that ?
>
> I use sofia/profile/user%domain^XXXXX to modify the SIP TO in the  
> INVITE sent to a registered user.
> I was expecting the syntax: user/<user> ^XXXXXX to work also, but it  
> doesn't.
>
> Is that normal ?
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et  
> ?tablis ? l'intention exclusive de ses destinataires. Toute  
> utilisation ou diffusion non autoris?e est interdite. Tout message  
> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou  
> falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le  
> d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
>
> Le 21/05/2010 ? 19:08, Phillip Jones a ?crit :
>
>> Hmm...
>>
>> That is it. % vs @ is the answer.
>>
>> Use @ the call loops back through and brings my world and  
>> FreeSWITCH crashing down.
>>
>> Use % and all is well in the world and the call is routed.
>>
>> Obviously I need to understand this better.
>>
>> On Fri, May 21, 2010 at 12:40 PM, Kristian Kielhofner <kris at kriskinc.com 
>> > wrote:
>> AFAIK the % let's Freeswitch know it's a local domain and doesn't  
>> need to do
>> any DNS lookups, etc. While it was never really necessary to use %  
>> it made a
>> lot sense for various reasons.
>>
>>
>> --
>> Kristian Kielhofner
>> http://blog.krisk.org
>>
>> ----- Original Message -----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> <freeswitch-users-bounces at lists.freeswitch.org>
>> To: freeswitch-users at lists.freeswitch.org
>> <freeswitch-users at lists.freeswitch.org>
>> Sent: Fri May 21 12:07:17 2010
>> Subject: Re: [Freeswitch-users] Quick question on Originate through  
>> api
>>
>> not necessary ....
>>
>> ----- Original Message -----
>> From: "Peter Olsson" <peter.olsson at visionutveckling.se>
>> To: <freeswitch-users at lists.freeswitch.org>
>> Sent: Friday, May 21, 2010 11:52 AM
>> Subject: Re: [Freeswitch-users] Quick question on Originate through  
>> api
>>
>>
>> For sofia/internal/ you should use a % as a separator I think,  
>> instead of
>> @-sign.
>>
>> /Peter
>> ________________________________________
>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
>> [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Phillip  
>> Jones
>> [pjintheusa at gmail.com]
>> Skickat: den 21 maj 2010 17:41
>> Till: freeswitch-users at lists.freeswitch.org
>> ?mne: [Freeswitch-users] Quick question on Originate through api
>>
>> Hi there,
>>
>> In my mind - where 6775557767 is a registered user, the following  
>> two are
>> the same:
>>
>> apiResult = fsApi.Execute("originate",
>> string 
>> .Format 
>> ("{{ignore_early_media 
>> = 
>> false 
>> ,absolute_codec_string 
>> ='PCMU',origination_caller_id_number=5556667776}}user/6775557767
>> '
>>
>> apiResult = fsApi.Execute("originate",
>> string 
>> .Format("{{ignore_early_media=false,absolute_codec_string='PCMU',origination_caller_id_number=5556667776}}sofia/internal/6775557767 at 121.763.76.76
>> '
>>
>> i.e. user/ and sofia/internal are the same thing.
>>
>> However on my system the user/ works great but sofia/internal loops  
>> call
>> back through in the public context.
>>
>> Can someone tell me the difference between these options?
>>
>> What am I missing?
>>
>> Thanks
>>
>>
>> Phil
>>
>>
>>
>> !DSPAM:4bf6abb932934294689129!
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> -- 
> 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From stevendt at primrosebank.net  Sat May 22 01:21:56 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Sat, 22 May 2010 09:21:56 +0100
Subject: [Freeswitch-users] SVN Mirror
Message-ID: <B3171C43FAC94239822FAB3CD8DC2D7E@bp1.ad.bp.com>

Hi Folks - Mike ?

The SVN Mirror looks like it is a bit behind, an SVN update shows the last changes on the 13th May, is there a problem with the Mirror at the moment ?

How far behind the Git will the SVN be ?

regards
Dave
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From jason at jasonjgw.net  Sat May 22 03:09:33 2010
From: jason at jasonjgw.net (Jason White)
Date: Sat, 22 May 2010 20:09:33 +1000
Subject: [Freeswitch-users] SVN Mirror
In-Reply-To: <B3171C43FAC94239822FAB3CD8DC2D7E@bp1.ad.bp.com>
References: <B3171C43FAC94239822FAB3CD8DC2D7E@bp1.ad.bp.com>
Message-ID: <20100522100933.GA8450@jdc.jasonjgw.net>

Dave Stevenson <stevendt at primrosebank.net> wrote:
> The SVN Mirror looks like it is a bit behind, an SVN update shows the last changes on the 13th May, is there a problem with the Mirror at the moment ?

The last Git commit was on 21 May.
> 
> How far behind the Git will the SVN be ?

Why not just switch to the git repository? This will avoid any such problem.
(I deleted my copy of the svn tree soon after the transition to Git was
completed).



From stevendt at primrosebank.net  Sat May 22 03:41:34 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Sat, 22 May 2010 11:41:34 +0100
Subject: [Freeswitch-users] SVN Mirror
References: <B3171C43FAC94239822FAB3CD8DC2D7E@bp1.ad.bp.com>
	<20100522100933.GA8450@jdc.jasonjgw.net>
Message-ID: <D34CD729D65140D0B1ED3C9726488BE6@bp1.ad.bp.com>

Jason,

trying to do that now ! Not quite going according to plan, but still working 
on it

regards
Dave


----- Original Message ----- 
From: "Jason White" <jason at jasonjgw.net>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, May 22, 2010 11:09 AM
Subject: Re: [Freeswitch-users] SVN Mirror


> Dave Stevenson <stevendt at primrosebank.net> wrote:
>> The SVN Mirror looks like it is a bit behind, an SVN update shows the 
>> last changes on the 13th May, is there a problem with the Mirror at the 
>> moment ?
>
> The last Git commit was on 21 May.
>>
>> How far behind the Git will the SVN be ?
>
> Why not just switch to the git repository? This will avoid any such 
> problem.
> (I deleted my copy of the svn tree soon after the transition to Git was
> completed).
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From jason at jasonjgw.net  Sat May 22 04:13:09 2010
From: jason at jasonjgw.net (Jason White)
Date: Sat, 22 May 2010 21:13:09 +1000
Subject: [Freeswitch-users] SVN Mirror
In-Reply-To: <D34CD729D65140D0B1ED3C9726488BE6@bp1.ad.bp.com>
References: <B3171C43FAC94239822FAB3CD8DC2D7E@bp1.ad.bp.com>
	<20100522100933.GA8450@jdc.jasonjgw.net>
	<D34CD729D65140D0B1ED3C9726488BE6@bp1.ad.bp.com>
Message-ID: <20100522111309.GA8783@jdc.jasonjgw.net>

Dave Stevenson <stevendt at primrosebank.net> wrote:
> trying to do that now ! Not quite going according to plan, but still working 
> on it

Perhaps you should post details of this if you would like help with the
transition.

Under Linux, at least, it's easy: just install Git, clone the repository and
build FreeSWITCH.

In my case, there were some local changes to package-related files that I
wanted to preserve. I placed these in a separate branch, which can be
periodically merged with the master branch.



From stevendt at primrosebank.net  Sat May 22 06:40:45 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Sat, 22 May 2010 14:40:45 +0100
Subject: [Freeswitch-users] SVN Mirror
References: <B3171C43FAC94239822FAB3CD8DC2D7E@bp1.ad.bp.com><20100522100933.GA8450@jdc.jasonjgw.net><D34CD729D65140D0B1ED3C9726488BE6@bp1.ad.bp.com>
	<20100522111309.GA8783@jdc.jasonjgw.net>
Message-ID: <5F9E6D41F9C64AA599D5530C9F21DE00@bp1.ad.bp.com>

Jason,

it's not quite so easy under Windows - at least, not for me.

I'm trying to install TortoiseGit, in Windows, there is a dependency on 
mystic which needs to be installed first (I found that out AFTER I'd 
installed TortoiseGit - ooops).

I have uninstalled/reinstaleld both programs now and still can't get 
TortoiseGit to work - I guess I need some help from someone running 
TortoiseGit under windows. I have downloaded msysgit and it has built Git 
correctly, but I think that there's something not right in the linkage 
between TortoiseGit and Git. I get an error in Tortoise that says "git have 
not installed" [sic]

regards
Dave

----- Original Message ----- 
From: "Jason White" <jason at jasonjgw.net>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, May 22, 2010 12:13 PM
Subject: Re: [Freeswitch-users] SVN Mirror


> Dave Stevenson <stevendt at primrosebank.net> wrote:
>> trying to do that now ! Not quite going according to plan, but still 
>> working
>> on it
>
> Perhaps you should post details of this if you would like help with the
> transition.
>
> Under Linux, at least, it's easy: just install Git, clone the repository 
> and
> build FreeSWITCH.
>
> In my case, there were some local changes to package-related files that I
> wanted to preserve. I placed these in a separate branch, which can be
> periodically merged with the master branch.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From jan.berger at video24.no  Sat May 22 07:04:58 2010
From: jan.berger at video24.no (Jan Berger)
Date: Sat, 22 May 2010 16:04:58 +0200
Subject: [Freeswitch-users] SVN Mirror
In-Reply-To: <5F9E6D41F9C64AA599D5530C9F21DE00@bp1.ad.bp.com>
References: <B3171C43FAC94239822FAB3CD8DC2D7E@bp1.ad.bp.com><20100522100933.GA8450@jdc.jasonjgw.net><D34CD729D65140D0B1ED3C9726488BE6@bp1.ad.bp.com>
	<20100522111309.GA8783@jdc.jasonjgw.net>
	<5F9E6D41F9C64AA599D5530C9F21DE00@bp1.ad.bp.com>
Message-ID: <326A92F1557241BA8C21B8FA73FD217B@dell9400>

I did not have to much trouble, but I installed thee ordinary git first.

Also - I found TourtoiseGit a bit strange, so I had to use git bash (command
line interface" to get it working. 

Jan

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Dave
Stevenson
Sent: 22. mai 2010 15:41
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] SVN Mirror

Jason,

it's not quite so easy under Windows - at least, not for me.

I'm trying to install TortoiseGit, in Windows, there is a dependency on 
mystic which needs to be installed first (I found that out AFTER I'd 
installed TortoiseGit - ooops).

I have uninstalled/reinstaleld both programs now and still can't get 
TortoiseGit to work - I guess I need some help from someone running 
TortoiseGit under windows. I have downloaded msysgit and it has built Git 
correctly, but I think that there's something not right in the linkage 
between TortoiseGit and Git. I get an error in Tortoise that says "git have 
not installed" [sic]

regards
Dave

----- Original Message ----- 
From: "Jason White" <jason at jasonjgw.net>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, May 22, 2010 12:13 PM
Subject: Re: [Freeswitch-users] SVN Mirror


> Dave Stevenson <stevendt at primrosebank.net> wrote:
>> trying to do that now ! Not quite going according to plan, but still 
>> working
>> on it
>
> Perhaps you should post details of this if you would like help with the
> transition.
>
> Under Linux, at least, it's easy: just install Git, clone the repository 
> and
> build FreeSWITCH.
>
> In my case, there were some local changes to package-related files that I
> wanted to preserve. I placed these in a separate branch, which can be
> periodically merged with the master branch.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 


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



From stevendt at primrosebank.net  Sat May 22 09:07:43 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Sat, 22 May 2010 17:07:43 +0100
Subject: [Freeswitch-users] SVN Mirror
References: <B3171C43FAC94239822FAB3CD8DC2D7E@bp1.ad.bp.com><20100522100933.GA8450@jdc.jasonjgw.net><D34CD729D65140D0B1ED3C9726488BE6@bp1.ad.bp.com><20100522111309.GA8783@jdc.jasonjgw.net><5F9E6D41F9C64AA599D5530C9F21DE00@bp1.ad.bp.com>
	<326A92F1557241BA8C21B8FA73FD217B@dell9400>
Message-ID: <D701F9999818488781541CCCBE0EDC60@bp1.ad.bp.com>

Hi Jan,

thanks for the reply.

Can you tell me how you "bash"ed it please ? Git and the bash CLI are all 
new to me I'm afraid,

regards
Dave


----- Original Message ----- 
From: "Jan Berger" <jan.berger at video24.no>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, May 22, 2010 3:04 PM
Subject: Re: [Freeswitch-users] SVN Mirror


>I did not have to much trouble, but I installed thee ordinary git first.
>
> Also - I found TourtoiseGit a bit strange, so I had to use git bash 
> (command
> line interface" to get it working.
>
> Jan
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Dave
> Stevenson
> Sent: 22. mai 2010 15:41
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] SVN Mirror
>
> Jason,
>
> it's not quite so easy under Windows - at least, not for me.
>
> I'm trying to install TortoiseGit, in Windows, there is a dependency on
> mystic which needs to be installed first (I found that out AFTER I'd
> installed TortoiseGit - ooops).
>
> I have uninstalled/reinstaleld both programs now and still can't get
> TortoiseGit to work - I guess I need some help from someone running
> TortoiseGit under windows. I have downloaded msysgit and it has built Git
> correctly, but I think that there's something not right in the linkage
> between TortoiseGit and Git. I get an error in Tortoise that says "git 
> have
> not installed" [sic]
>
> regards
> Dave
>
> ----- Original Message ----- 
> From: "Jason White" <jason at jasonjgw.net>
> To: <freeswitch-users at lists.freeswitch.org>
> Sent: Saturday, May 22, 2010 12:13 PM
> Subject: Re: [Freeswitch-users] SVN Mirror
>
>
>> Dave Stevenson <stevendt at primrosebank.net> wrote:
>>> trying to do that now ! Not quite going according to plan, but still
>>> working
>>> on it
>>
>> Perhaps you should post details of this if you would like help with the
>> transition.
>>
>> Under Linux, at least, it's easy: just install Git, clone the repository
>> and
>> build FreeSWITCH.
>>
>> In my case, there were some local changes to package-related files that I
>> wanted to preserve. I placed these in a separate branch, which can be
>> periodically merged with the master branch.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From stevendt at primrosebank.net  Sat May 22 10:22:02 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Sat, 22 May 2010 18:22:02 +0100
Subject: [Freeswitch-users] SVN Mirror - Closed
References: <B3171C43FAC94239822FAB3CD8DC2D7E@bp1.ad.bp.com><20100522100933.GA8450@jdc.jasonjgw.net><D34CD729D65140D0B1ED3C9726488BE6@bp1.ad.bp.com><20100522111309.GA8783@jdc.jasonjgw.net><5F9E6D41F9C64AA599D5530C9F21DE00@bp1.ad.bp.com><326A92F1557241BA8C21B8FA73FD217B@dell9400>
	<D701F9999818488781541CCCBE0EDC60@bp1.ad.bp.com>
Message-ID: <7B8D2674FE3349708310970BEA69EF97@bp1.ad.bp.com>

Thanks guys,

I think that I've got it now - ended up removing msysgit and installing 
another version (actually, a slightly older one!).

That seems to have fixed it now,

regards
Dave


----- Original Message ----- 
From: "Dave Stevenson" <stevendt at primrosebank.net>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, May 22, 2010 5:07 PM
Subject: Re: [Freeswitch-users] SVN Mirror


> Hi Jan,
>
> thanks for the reply.
>
> Can you tell me how you "bash"ed it please ? Git and the bash CLI are all
> new to me I'm afraid,
>
> regards
> Dave
>
>
> ----- Original Message ----- 
> From: "Jan Berger" <jan.berger at video24.no>
> To: <freeswitch-users at lists.freeswitch.org>
> Sent: Saturday, May 22, 2010 3:04 PM
> Subject: Re: [Freeswitch-users] SVN Mirror
>
>
>>I did not have to much trouble, but I installed thee ordinary git first.
>>
>> Also - I found TourtoiseGit a bit strange, so I had to use git bash
>> (command
>> line interface" to get it working.
>>
>> Jan
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Dave
>> Stevenson
>> Sent: 22. mai 2010 15:41
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] SVN Mirror
>>
>> Jason,
>>
>> it's not quite so easy under Windows - at least, not for me.
>>
>> I'm trying to install TortoiseGit, in Windows, there is a dependency on
>> mystic which needs to be installed first (I found that out AFTER I'd
>> installed TortoiseGit - ooops).
>>
>> I have uninstalled/reinstaleld both programs now and still can't get
>> TortoiseGit to work - I guess I need some help from someone running
>> TortoiseGit under windows. I have downloaded msysgit and it has built Git
>> correctly, but I think that there's something not right in the linkage
>> between TortoiseGit and Git. I get an error in Tortoise that says "git
>> have
>> not installed" [sic]
>>
>> regards
>> Dave
>>
>> ----- Original Message ----- 
>> From: "Jason White" <jason at jasonjgw.net>
>> To: <freeswitch-users at lists.freeswitch.org>
>> Sent: Saturday, May 22, 2010 12:13 PM
>> Subject: Re: [Freeswitch-users] SVN Mirror
>>
>>
>>> Dave Stevenson <stevendt at primrosebank.net> wrote:
>>>> trying to do that now ! Not quite going according to plan, but still
>>>> working
>>>> on it
>>>
>>> Perhaps you should post details of this if you would like help with the
>>> transition.
>>>
>>> Under Linux, at least, it's easy: just install Git, clone the repository
>>> and
>>> build FreeSWITCH.
>>>
>>> In my case, there were some local changes to package-related files that 
>>> I
>>> wanted to preserve. I placed these in a separate branch, which can be
>>> periodically merged with the master branch.
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From stevendt at primrosebank.net  Sat May 22 11:05:48 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Sat, 22 May 2010 19:05:48 +0100
Subject: [Freeswitch-users] Mod_skinny: Call for testing!
References: <AANLkTik8hHRtOci8qTXmZVHOWvVxRg-OPm5zyRAMrWEe@mail.gmail.com><F32D0E59CC994113BC88C7902AFC87BF@bp1.ad.bp.com>,
	<AANLkTikJ1biYWWY-M4qYMWgsYfGelxvFu91EVnBqDlcq@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C557DE57F78@cooper>
Message-ID: <3FDDF9772BE64AB8A81243DD375E46BC@bp1.ad.bp.com>

Peter,

after some hiccups, I now have Git installed and just downloaded the latest 
Git (21/05/10).

I don't see modskinny in the Visual Studio 2008 Express Solution - how are 
you building it ?

regards
Dave


----- Original Message ----- 
From: "Peter Olsson" <peter.olsson at visionutveckling.se>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Friday, May 21, 2010 4:16 PM
Subject: Re: [Freeswitch-users] Mod_skinny: Call for testing!


I tried building today, and it builds on Windows on latest git - just needs 
some testing now :)

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org 
[freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Mathieu Parent 
[math.parent at gmail.com]
Skickat: den 21 maj 2010 16:24
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Mod_skinny: Call for testing!

On Fri, May 21, 2010 at 3:46 PM, Dave Stevenson
<stevendt at primrosebank.net> wrote:
> Mathieu,
>
> just updated to 17539 (SVN) and skinny is not included in the Windows 
> Build
> Solution to VS2008
>
> Does it need to be compiled & linked outside of the VS Build ?

SVN is out of sync, git has Windows support since yesterday:

commit bfb6a8b1f5f835008d1e0fbd90d2e120814f341a
Author: Mathieu Parent <math.parent at gmail.com>
Date:   Thu May 20 23:19:10 2010 +0200

    Skinny: Initial Windows support

    Thanks to Peter Olsson

    Closes: MODSKINNY-3

I have not been able to test it, you can comment MODSKINNY-3 if you
still have problem after the patch applied.

Mathieu Parent

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

!DSPAM:4bf6994c32931601618424!


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



From peter.olsson at visionutveckling.se  Sat May 22 11:14:53 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sat, 22 May 2010 20:14:53 +0200
Subject: [Freeswitch-users] Mod_skinny: Call for testing!
In-Reply-To: <3FDDF9772BE64AB8A81243DD375E46BC@bp1.ad.bp.com>
References: <AANLkTik8hHRtOci8qTXmZVHOWvVxRg-OPm5zyRAMrWEe@mail.gmail.com><F32D0E59CC994113BC88C7902AFC87BF@bp1.ad.bp.com>,
	<AANLkTikJ1biYWWY-M4qYMWgsYfGelxvFu91EVnBqDlcq@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C557DE57F78@cooper>,
	<3FDDF9772BE64AB8A81243DD375E46BC@bp1.ad.bp.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C557DE57F86@cooper>

Hi Dave,

Actually, the patch was just made for Visual Studio, not the Express version. But I think you should be able to figure it out by comparing Freeswitch.2008.sln to Freeswitch.2008.express.sln.

I will try to make some time to get it fixed for express version during the next week.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Dave Stevenson [stevendt at primrosebank.net]
Skickat: den 22 maj 2010 20:05
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Mod_skinny: Call for testing!

Peter,

after some hiccups, I now have Git installed and just downloaded the latest
Git (21/05/10).

I don't see modskinny in the Visual Studio 2008 Express Solution - how are
you building it ?

regards
Dave


----- Original Message -----
From: "Peter Olsson" <peter.olsson at visionutveckling.se>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Friday, May 21, 2010 4:16 PM
Subject: Re: [Freeswitch-users] Mod_skinny: Call for testing!


I tried building today, and it builds on Windows on latest git - just needs
some testing now :)

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org
[freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Mathieu Parent
[math.parent at gmail.com]
Skickat: den 21 maj 2010 16:24
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Mod_skinny: Call for testing!

On Fri, May 21, 2010 at 3:46 PM, Dave Stevenson
<stevendt at primrosebank.net> wrote:
> Mathieu,
>
> just updated to 17539 (SVN) and skinny is not included in the Windows
> Build
> Solution to VS2008
>
> Does it need to be compiled & linked outside of the VS Build ?

SVN is out of sync, git has Windows support since yesterday:

commit bfb6a8b1f5f835008d1e0fbd90d2e120814f341a
Author: Mathieu Parent <math.parent at gmail.com>
Date:   Thu May 20 23:19:10 2010 +0200

    Skinny: Initial Windows support

    Thanks to Peter Olsson

    Closes: MODSKINNY-3

I have not been able to test it, you can comment MODSKINNY-3 if you
still have problem after the patch applied.

Mathieu Parent

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




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


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

!DSPAM:4bf81edf32932581712320!



From ingduquerueda at gmail.com  Fri May 21 16:50:25 2010
From: ingduquerueda at gmail.com (Alejandro Duque Rueda)
Date: Fri, 21 May 2010 18:50:25 -0500
Subject: [Freeswitch-users] confirm
	155a8a12b77a3382db745b073f0d8f6a37c95f5d
In-Reply-To: <mailman.0.1274485471.6968.freeswitch-users@lists.freeswitch.org>
References: <mailman.0.1274485471.6968.freeswitch-users@lists.freeswitch.org>
Message-ID: <AANLkTimf4CAAmI0hh4MlYe1bbRhVAfO5L_hkF5cNjoGz@mail.gmail.com>

Hello, I am working with freeswitch on Open wrt. I have 2 modules that
doesn't exists information in the wiki: mod_portaudio_stream and mod_alsa.
Someone can tell me for what are those modules, and if someone have worked
with them. Thank you
-------------- next part --------------
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From romerocarlos24 at gmail.com  Fri May 21 19:59:14 2010
From: romerocarlos24 at gmail.com (Carlos Romero)
Date: Fri, 21 May 2010 22:59:14 -0400
Subject: [Freeswitch-users] mod_dptools/redirect and 300 Multiple Choices
In-Reply-To: <AANLkTikfqwv_3UxrKOqQTKaE518ENhA5zw7Usoj3vH_G@mail.gmail.com>
References: <AANLkTinYtV2CL1H_hME51kP4OpJLaYeuQK0jUEnrimAz@mail.gmail.com>
	<AANLkTikfqwv_3UxrKOqQTKaE518ENhA5zw7Usoj3vH_G@mail.gmail.com>
Message-ID: <AANLkTikmqhQN5lRLKUZdvq-78gFEOoQpihLxx0yOsnh6@mail.gmail.com>

why choose  a library that is hard to work with in the first place?

it seems like sofia has many downsides... limited threads per
sip-profile and now this

On Fri, May 21, 2010 at 3:20 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Thu, May 20, 2010 at 11:46 PM, Carlos Romero <romerocarlos24 at gmail.com>wrote:
>
>> what do you mean that you don't have access to it?
>>
>> you have the code
>>
>
> The SIP stack is supplied by Nokia. Open source or not, it is bad form to
> muck with someone else's library without due cause. Like all robust SIP
> stacks, Sofia is huge, deep, complex, and otherwise uninviting of those who
> would waltz in and do whatever they wish. If you feel that it is even within
> the realm of possibility then I recommend that you find the location in the
> Sofia library where this takes place, make the appropriate changes, and
> thoroughly test for regressions. Submit the patch to the Sofia devs who will
> then evaluate it and see if it is worthy of inclusion.
>
> Or more simply put: "having the code" and "having access" are COMPLETELY
> different concepts.
> -MC
>
>
>>
>> On Thu, May 20, 2010 at 3:23 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>> > not without intervention from the sofia devs,
>> > Sofia sip does that deep down and we don't have access to it.
>> > even when we supply it as a one big string it renders it out depending
>> on
>> > those settings.
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

On Fri, May 21, 2010 at 4:34 AM, Carlos Romero <romerocarlos24 at gmail.com> wrote:
> what do you mean that you don't have access to it?
>
> you have the code
>
> On Thu, May 20, 2010 at 3:23 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>> not without intervention from the sofia devs,
>> Sofia sip does that deep down and we don't have access to it.
>> even when we supply it as a one big string it renders it out depending on
>> those settings.
>>
>


From mike at jerris.com  Sat May 22 12:46:48 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 22 May 2010 15:46:48 -0400
Subject: [Freeswitch-users] Conference command "relate", nospeak/nohear
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C557DE57F81@cooper>
References: <549CFEF87AEDE841A38E9D15EAB4C04C557DE57F81@cooper>
Message-ID: <C79F62B0-55F1-49D8-B530-CCA3C85A8179@jerris.com>

Yes.

Mike

On May 22, 2010, at 2:41 AM, Peter Olsson wrote:

> Hello everyone,
> 
> I have a small question about the mod_conference relate command. What's the expected behaviour of "nohear"? Is it supposed to work as nospeak, but in the other direction?
> 
> When running "conference xx relate member_x member_y nospeak", the audio from member_x to member_y is muted, but when using nohear I didn't notice any difference at all (I could still hear audio everywhere). Is it supposed to work the opposite way, so when using nospeak|nohear as parameter, the audio from x to y is muted, and also the other way around?



From sos at sokhapkin.dyndns.org  Sat May 22 12:53:42 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Sat, 22 May 2010 15:53:42 -0400
Subject: [Freeswitch-users] mod_dptools/redirect and 300 Multiple Choices
In-Reply-To: <AANLkTikmqhQN5lRLKUZdvq-78gFEOoQpihLxx0yOsnh6@mail.gmail.com>
References: <AANLkTinYtV2CL1H_hME51kP4OpJLaYeuQK0jUEnrimAz@mail.gmail.com>
	<AANLkTikfqwv_3UxrKOqQTKaE518ENhA5zw7Usoj3vH_G@mail.gmail.com>
	<AANLkTikmqhQN5lRLKUZdvq-78gFEOoQpihLxx0yOsnh6@mail.gmail.com>
Message-ID: <201005221553.42315.sos@sokhapkin.dyndns.org>

Could you suggest a better open source SIP implementation?

On Friday 21 May 2010, Carlos Romero wrote:
> why choose  a library that is hard to work with in the first place?
> 
> it seems like sofia has many downsides... limited threads per
> sip-profile and now this
> 
> On Fri, May 21, 2010 at 3:20 PM, Michael Collins <msc at freeswitch.org> 
wrote:
> > On Thu, May 20, 2010 at 11:46 PM, Carlos Romero <romerocarlos24 at 
gmail.com>wrote:
> >> what do you mean that you don't have access to it?
> >>
> >> you have the code
> >
> > The SIP stack is supplied by Nokia. Open source or not, it is bad form to
> > muck with someone else's library without due cause. Like all robust SIP
> > stacks, Sofia is huge, deep, complex, and otherwise uninviting of those
> > who would waltz in and do whatever they wish. If you feel that it is even
> > within the realm of possibility then I recommend that you find the
> > location in the Sofia library where this takes place, make the
> > appropriate changes, and thoroughly test for regressions. Submit the
> > patch to the Sofia devs who will then evaluate it and see if it is worthy
> > of inclusion.
> >
> > Or more simply put: "having the code" and "having access" are COMPLETELY
> > different concepts.
> > -MC
> >
> >> On Thu, May 20, 2010 at 3:23 PM, Anthony Minessale
> >>
> >> <anthony.minessale at gmail.com> wrote:
> >> > not without intervention from the sofia devs,
> >> > Sofia sip does that deep down and we don't have access to it.
> >> > even when we supply it as a one big string it renders it out depending
> >>
> >> on
> >>
> >> > those settings.
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> On Fri, May 21, 2010 at 4:34 AM, Carlos Romero <romerocarlos24 at gmail.com> 
wrote:
> > what do you mean that you don't have access to it?
> >
> > you have the code
> >
> > On Thu, May 20, 2010 at 3:23 PM, Anthony Minessale
> >
> > <anthony.minessale at gmail.com> wrote:
> >> not without intervention from the sofia devs,
> >> Sofia sip does that deep down and we don't have access to it.
> >> even when we supply it as a one big string it renders it out depending
> >> on those settings.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From mike at jerris.com  Sat May 22 13:50:59 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 22 May 2010 16:50:59 -0400
Subject: [Freeswitch-users] confirm
	155a8a12b77a3382db745b073f0d8f6a37c95f5d
In-Reply-To: <AANLkTimf4CAAmI0hh4MlYe1bbRhVAfO5L_hkF5cNjoGz@mail.gmail.com>
References: <mailman.0.1274485471.6968.freeswitch-users@lists.freeswitch.org>
	<AANLkTimf4CAAmI0hh4MlYe1bbRhVAfO5L_hkF5cNjoGz@mail.gmail.com>
Message-ID: <6085DE86-CA07-42EB-B635-28C467A040E6@jerris.com>

These modules shouldn't be that useful for embedded devices without audio hardware.  mod_alsa is support for alsa specifically for crippled alsa setups such as the n800, usually you could just use mod_portaudio, which is probably not even useful for wrt.  mod_portaudio_stream lets you create a stream interface, such as for music on hold, from audio devices like a sound card, again, probably not useful for wrt.

Mike


On May 21, 2010, at 7:50 PM, Alejandro Duque Rueda wrote:

> Hello, I am working with freeswitch on Open wrt. I have 2 modules that doesn't exists information in the wiki: mod_portaudio_stream and mod_alsa. Someone can tell me for what are those modules, and if someone have worked with them. Thank you



From mike at jerris.com  Sat May 22 13:52:33 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 22 May 2010 16:52:33 -0400
Subject: [Freeswitch-users] mod_dptools/redirect and 300 Multiple Choices
In-Reply-To: <AANLkTikmqhQN5lRLKUZdvq-78gFEOoQpihLxx0yOsnh6@mail.gmail.com>
References: <AANLkTinYtV2CL1H_hME51kP4OpJLaYeuQK0jUEnrimAz@mail.gmail.com>
	<AANLkTikfqwv_3UxrKOqQTKaE518ENhA5zw7Usoj3vH_G@mail.gmail.com>
	<AANLkTikmqhQN5lRLKUZdvq-78gFEOoQpihLxx0yOsnh6@mail.gmail.com>
Message-ID: <2C75D7EF-120A-44A6-89CB-243A34B67D13@jerris.com>

Thank you for your offer to create a new open source supported sip library that is better than every one we have tried over the past 6 years.  I look forward to your contribution.

Mike

On May 21, 2010, at 10:59 PM, Carlos Romero wrote:

> why choose  a library that is hard to work with in the first place?
> 
> it seems like sofia has many downsides... limited threads per
> sip-profile and now this
> 



From stevendt at primrosebank.net  Sat May 22 14:01:11 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Sat, 22 May 2010 22:01:11 +0100
Subject: [Freeswitch-users] Mod_skinny: Call for testing!
References: <AANLkTik8hHRtOci8qTXmZVHOWvVxRg-OPm5zyRAMrWEe@mail.gmail.com><F32D0E59CC994113BC88C7902AFC87BF@bp1.ad.bp.com>,
	<AANLkTikJ1biYWWY-M4qYMWgsYfGelxvFu91EVnBqDlcq@mail.gmail.com><549CFEF87AEDE841A38E9D15EAB4C04C557DE57F78@cooper>,
	<3FDDF9772BE64AB8A81243DD375E46BC@bp1.ad.bp.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C557DE57F86@cooper>
Message-ID: <CB226C919BE944B59B0934AB1C644169@bp1.ad.bp.com>

Hi Peter,

thanks a lot - I'd appreciate it if you can take a look when you get time - 
no great hurry. I have a couple of sccp phones that I'd like to get working 
but there's no real urgency.

There's something quite odd going on though - well, I don't understand it !

Just looked in the VS2008 Solution file and see a reference to 
mod_skinny.vcproj - there's no matching file in my git download ?

regards
Dave



----- Original Message ----- 
From: "Peter Olsson" <peter.olsson at visionutveckling.se>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, May 22, 2010 7:14 PM
Subject: Re: [Freeswitch-users] Mod_skinny: Call for testing!


Hi Dave,

Actually, the patch was just made for Visual Studio, not the Express 
version. But I think you should be able to figure it out by comparing 
Freeswitch.2008.sln to Freeswitch.2008.express.sln.

I will try to make some time to get it fixed for express version during the 
next week.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org 
[freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Dave Stevenson 
[stevendt at primrosebank.net]
Skickat: den 22 maj 2010 20:05
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Mod_skinny: Call for testing!

Peter,

after some hiccups, I now have Git installed and just downloaded the latest
Git (21/05/10).

I don't see modskinny in the Visual Studio 2008 Express Solution - how are
you building it ?

regards
Dave


----- Original Message -----
From: "Peter Olsson" <peter.olsson at visionutveckling.se>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Friday, May 21, 2010 4:16 PM
Subject: Re: [Freeswitch-users] Mod_skinny: Call for testing!


I tried building today, and it builds on Windows on latest git - just needs
some testing now :)

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org
[freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Mathieu Parent
[math.parent at gmail.com]
Skickat: den 21 maj 2010 16:24
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Mod_skinny: Call for testing!

On Fri, May 21, 2010 at 3:46 PM, Dave Stevenson
<stevendt at primrosebank.net> wrote:
> Mathieu,
>
> just updated to 17539 (SVN) and skinny is not included in the Windows
> Build
> Solution to VS2008
>
> Does it need to be compiled & linked outside of the VS Build ?

SVN is out of sync, git has Windows support since yesterday:

commit bfb6a8b1f5f835008d1e0fbd90d2e120814f341a
Author: Mathieu Parent <math.parent at gmail.com>
Date:   Thu May 20 23:19:10 2010 +0200

    Skinny: Initial Windows support

    Thanks to Peter Olsson

    Closes: MODSKINNY-3

I have not been able to test it, you can comment MODSKINNY-3 if you
still have problem after the patch applied.

Mathieu Parent

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




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


_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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!DSPAM:4bf81edf32932581712320!


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From peter.olsson at visionutveckling.se  Sat May 22 15:19:09 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sun, 23 May 2010 00:19:09 +0200
Subject: [Freeswitch-users] Mod_skinny: Call for testing!
In-Reply-To: <CB226C919BE944B59B0934AB1C644169@bp1.ad.bp.com>
References: <AANLkTik8hHRtOci8qTXmZVHOWvVxRg-OPm5zyRAMrWEe@mail.gmail.com><F32D0E59CC994113BC88C7902AFC87BF@bp1.ad.bp.com>,
	<AANLkTikJ1biYWWY-M4qYMWgsYfGelxvFu91EVnBqDlcq@mail.gmail.com><549CFEF87AEDE841A38E9D15EAB4C04C557DE57F78@cooper>,
	<3FDDF9772BE64AB8A81243DD375E46BC@bp1.ad.bp.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C557DE57F86@cooper>,
	<CB226C919BE944B59B0934AB1C644169@bp1.ad.bp.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C557DE57F87@cooper>

Dave,

It seems like it was never added, I've asked Mathieu to add this file as well. I never noticed, since I already had my own local copy... If you want to, it's possible to download it directly from this link: http://jira.freeswitch.org/browse/MODSKINNY-3.

/Peter

________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Dave Stevenson [stevendt at primrosebank.net]
Skickat: den 22 maj 2010 23:01
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Mod_skinny: Call for testing!

Hi Peter,

thanks a lot - I'd appreciate it if you can take a look when you get time -
no great hurry. I have a couple of sccp phones that I'd like to get working
but there's no real urgency.

There's something quite odd going on though - well, I don't understand it !

Just looked in the VS2008 Solution file and see a reference to
mod_skinny.vcproj - there's no matching file in my git download ?

regards
Dave



----- Original Message -----
From: "Peter Olsson" <peter.olsson at visionutveckling.se>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, May 22, 2010 7:14 PM
Subject: Re: [Freeswitch-users] Mod_skinny: Call for testing!


Hi Dave,

Actually, the patch was just made for Visual Studio, not the Express
version. But I think you should be able to figure it out by comparing
Freeswitch.2008.sln to Freeswitch.2008.express.sln.

I will try to make some time to get it fixed for express version during the
next week.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org
[freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Dave Stevenson
[stevendt at primrosebank.net]
Skickat: den 22 maj 2010 20:05
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Mod_skinny: Call for testing!

Peter,

after some hiccups, I now have Git installed and just downloaded the latest
Git (21/05/10).

I don't see modskinny in the Visual Studio 2008 Express Solution - how are
you building it ?

regards
Dave


----- Original Message -----
From: "Peter Olsson" <peter.olsson at visionutveckling.se>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Friday, May 21, 2010 4:16 PM
Subject: Re: [Freeswitch-users] Mod_skinny: Call for testing!


I tried building today, and it builds on Windows on latest git - just needs
some testing now :)

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org
[freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Mathieu Parent
[math.parent at gmail.com]
Skickat: den 21 maj 2010 16:24
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Mod_skinny: Call for testing!

On Fri, May 21, 2010 at 3:46 PM, Dave Stevenson
<stevendt at primrosebank.net> wrote:
> Mathieu,
>
> just updated to 17539 (SVN) and skinny is not included in the Windows
> Build
> Solution to VS2008
>
> Does it need to be compiled & linked outside of the VS Build ?

SVN is out of sync, git has Windows support since yesterday:

commit bfb6a8b1f5f835008d1e0fbd90d2e120814f341a
Author: Mathieu Parent <math.parent at gmail.com>
Date:   Thu May 20 23:19:10 2010 +0200

    Skinny: Initial Windows support

    Thanks to Peter Olsson

    Closes: MODSKINNY-3

I have not been able to test it, you can comment MODSKINNY-3 if you
still have problem after the patch applied.

Mathieu Parent

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




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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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!DSPAM:4bf847d432933212913465!



From mike at jerris.com  Sat May 22 15:25:27 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 22 May 2010 18:25:27 -0400
Subject: [Freeswitch-users] Mod_skinny: Call for testing!
In-Reply-To: <3A8B4A19CB3C4270B20D9980B9E02F45@bp1.ad.bp.com>
References: <AANLkTik8hHRtOci8qTXmZVHOWvVxRg-OPm5zyRAMrWEe@mail.gmail.com><F32D0E59CC994113BC88C7902AFC87BF@bp1.ad.bp.com>,
	<AANLkTikJ1biYWWY-M4qYMWgsYfGelxvFu91EVnBqDlcq@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C557DE57F78@cooper>
	<3A8B4A19CB3C4270B20D9980B9E02F45@bp1.ad.bp.com>
Message-ID: <D7473B43-1129-4FDD-B956-D6430C562C64@jerris.com>

It should be updated now.

Mike

On May 21, 2010, at 12:52 PM, Dave Stevenson wrote:

> Hi,
> 
> thanks Peter.
> 
> I did not realise that the Windows build file was so new, OK, I guess that 
> I'll just have to wait for the SVN to catch up. I don't really want to 
> update to Git at the moment,
> 
> regards
> Dave
> 



From mike at jerris.com  Sat May 22 16:19:53 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 22 May 2010 19:19:53 -0400
Subject: [Freeswitch-users] Mod_skinny: Call for testing!
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C557DE57F87@cooper>
References: <AANLkTik8hHRtOci8qTXmZVHOWvVxRg-OPm5zyRAMrWEe@mail.gmail.com><F32D0E59CC994113BC88C7902AFC87BF@bp1.ad.bp.com>,
	<AANLkTikJ1biYWWY-M4qYMWgsYfGelxvFu91EVnBqDlcq@mail.gmail.com><549CFEF87AEDE841A38E9D15EAB4C04C557DE57F78@cooper>,
	<3FDDF9772BE64AB8A81243DD375E46BC@bp1.ad.bp.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C557DE57F86@cooper>,
	<CB226C919BE944B59B0934AB1C644169@bp1.ad.bp.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C557DE57F87@cooper>
Message-ID: <4113C83C-8F62-4258-B8AC-BC55AB9B8B08@jerris.com>

[master f89cbdd] add mod_skinny vcproj file for windows build (MODSKINNY-3)

On May 22, 2010, at 6:19 PM, Peter Olsson wrote:

> Dave,
> 
> It seems like it was never added, I've asked Mathieu to add this file as well. I never noticed, since I already had my own local copy... If you want to, it's possible to download it directly from this link: http://jira.freeswitch.org/browse/MODSKINNY-3.



From romerocarlos24 at gmail.com  Sat May 22 17:27:41 2010
From: romerocarlos24 at gmail.com (Carlos Romero)
Date: Sat, 22 May 2010 20:27:41 -0400
Subject: [Freeswitch-users] mod_dptools/redirect and 300 Multiple Choices
In-Reply-To: <201005221553.42315.sos@sokhapkin.dyndns.org>
References: <AANLkTinYtV2CL1H_hME51kP4OpJLaYeuQK0jUEnrimAz@mail.gmail.com>
	<AANLkTikfqwv_3UxrKOqQTKaE518ENhA5zw7Usoj3vH_G@mail.gmail.com>
	<AANLkTikmqhQN5lRLKUZdvq-78gFEOoQpihLxx0yOsnh6@mail.gmail.com>
	<201005221553.42315.sos@sokhapkin.dyndns.org>
Message-ID: <AANLkTinJ04EHUyvEw8sv81DC3HR7Chd7FywxZVpVhkP6@mail.gmail.com>

http://www.pjsip.org/

On Sat, May 22, 2010 at 3:53 PM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org> wrote:
> Could you suggest a better open source SIP implementation?
>
> On Friday 21 May 2010, Carlos Romero wrote:
>> why choose ?a library that is hard to work with in the first place?
>>
>> it seems like sofia has many downsides... limited threads per
>> sip-profile and now this
>>
>> On Fri, May 21, 2010 at 3:20 PM, Michael Collins <msc at freeswitch.org>
> wrote:
>> > On Thu, May 20, 2010 at 11:46 PM, Carlos Romero <romerocarlos24 at
> gmail.com>wrote:
>> >> what do you mean that you don't have access to it?
>> >>
>> >> you have the code
>> >
>> > The SIP stack is supplied by Nokia. Open source or not, it is bad form to
>> > muck with someone else's library without due cause. Like all robust SIP
>> > stacks, Sofia is huge, deep, complex, and otherwise uninviting of those
>> > who would waltz in and do whatever they wish. If you feel that it is even
>> > within the realm of possibility then I recommend that you find the
>> > location in the Sofia library where this takes place, make the
>> > appropriate changes, and thoroughly test for regressions. Submit the
>> > patch to the Sofia devs who will then evaluate it and see if it is worthy
>> > of inclusion.
>> >
>> > Or more simply put: "having the code" and "having access" are COMPLETELY
>> > different concepts.
>> > -MC
>> >
>> >> On Thu, May 20, 2010 at 3:23 PM, Anthony Minessale
>> >>
>> >> <anthony.minessale at gmail.com> wrote:
>> >> > not without intervention from the sofia devs,
>> >> > Sofia sip does that deep down and we don't have access to it.
>> >> > even when we supply it as a one big string it renders it out depending
>> >>
>> >> on
>> >>
>> >> > those settings.
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> On Fri, May 21, 2010 at 4:34 AM, Carlos Romero <romerocarlos24 at gmail.com>
> wrote:
>> > what do you mean that you don't have access to it?
>> >
>> > you have the code
>> >
>> > On Thu, May 20, 2010 at 3:23 PM, Anthony Minessale
>> >
>> > <anthony.minessale at gmail.com> wrote:
>> >> not without intervention from the sofia devs,
>> >> Sofia sip does that deep down and we don't have access to it.
>> >> even when we supply it as a one big string it renders it out depending
>> >> on those settings.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>


From mike at jerris.com  Sat May 22 17:45:29 2010
From: mike at jerris.com (Michael Jerris)
Date: Sat, 22 May 2010 20:45:29 -0400
Subject: [Freeswitch-users] mod_dptools/redirect and 300 Multiple Choices
In-Reply-To: <AANLkTinJ04EHUyvEw8sv81DC3HR7Chd7FywxZVpVhkP6@mail.gmail.com>
References: <AANLkTinYtV2CL1H_hME51kP4OpJLaYeuQK0jUEnrimAz@mail.gmail.com>
	<AANLkTikfqwv_3UxrKOqQTKaE518ENhA5zw7Usoj3vH_G@mail.gmail.com>
	<AANLkTikmqhQN5lRLKUZdvq-78gFEOoQpihLxx0yOsnh6@mail.gmail.com>
	<201005221553.42315.sos@sokhapkin.dyndns.org>
	<AANLkTinJ04EHUyvEw8sv81DC3HR7Chd7FywxZVpVhkP6@mail.gmail.com>
Message-ID: <9122C13D-91BC-4AC4-AAE7-18D40182FCAE@jerris.com>

Been there, done that.  I was hoping my comments would make this clear, but apparently not.  We have tried EVERY sip library out there.  We stuck with sofia because it was by far the best one out there for our needs.  pjsip had significant thread safety issues, we tried for many many months to resolve them with the primary developer, and failed.  Sofia also outperforms pjsip.  This is all aside of the licensing issues with pjsip.

Mike

On May 22, 2010, at 8:27 PM, Carlos Romero wrote:

> http://www.pjsip.org/
> 
> On Sat, May 22, 2010 at 3:53 PM, Sergey Okhapkin
> <sos at sokhapkin.dyndns.org> wrote:
>> Could you suggest a better open source SIP implementation?
>> 



From sos at sokhapkin.dyndns.org  Sat May 22 18:11:30 2010
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Sat, 22 May 2010 21:11:30 -0400
Subject: [Freeswitch-users] mod_dptools/redirect and 300 Multiple Choices
In-Reply-To: <AANLkTinJ04EHUyvEw8sv81DC3HR7Chd7FywxZVpVhkP6@mail.gmail.com>
References: <AANLkTinYtV2CL1H_hME51kP4OpJLaYeuQK0jUEnrimAz@mail.gmail.com>
	<201005221553.42315.sos@sokhapkin.dyndns.org>
	<AANLkTinJ04EHUyvEw8sv81DC3HR7Chd7FywxZVpVhkP6@mail.gmail.com>
Message-ID: <201005222111.30178.sos@sokhapkin.dyndns.org>

http://www.pjsip.org/apps.htm doesn't list any serious server side software 
based on pjsip, but simple client side implementations only. Perhaps there is 
a reason for that...

On Saturday 22 May 2010, Carlos Romero wrote:
> http://www.pjsip.org/
> 
> On Sat, May 22, 2010 at 3:53 PM, Sergey Okhapkin
> 
> <sos at sokhapkin.dyndns.org> wrote:
> > Could you suggest a better open source SIP implementation?
> >
> > On Friday 21 May 2010, Carlos Romero wrote:
> >> why choose  a library that is hard to work with in the first place?
> >>
> >> it seems like sofia has many downsides... limited threads per
> >> sip-profile and now this
> >>
> >> On Fri, May 21, 2010 at 3:20 PM, Michael Collins <msc at freeswitch.org>
> >
> > wrote:
> >> > On Thu, May 20, 2010 at 11:46 PM, Carlos Romero <romerocarlos24 at
> >
> > gmail.com>wrote:
> >> >> what do you mean that you don't have access to it?
> >> >>
> >> >> you have the code
> >> >
> >> > The SIP stack is supplied by Nokia. Open source or not, it is bad form
> >> > to muck with someone else's library without due cause. Like all robust
> >> > SIP stacks, Sofia is huge, deep, complex, and otherwise uninviting of
> >> > those who would waltz in and do whatever they wish. If you feel that
> >> > it is even within the realm of possibility then I recommend that you
> >> > find the location in the Sofia library where this takes place, make
> >> > the appropriate changes, and thoroughly test for regressions. Submit
> >> > the patch to the Sofia devs who will then evaluate it and see if it is
> >> > worthy of inclusion.
> >> >
> >> > Or more simply put: "having the code" and "having access" are
> >> > COMPLETELY different concepts.
> >> > -MC
> >> >
> >> >> On Thu, May 20, 2010 at 3:23 PM, Anthony Minessale
> >> >>
> >> >> <anthony.minessale at gmail.com> wrote:
> >> >> > not without intervention from the sofia devs,
> >> >> > Sofia sip does that deep down and we don't have access to it.
> >> >> > even when we supply it as a one big string it renders it out
> >> >> > depending
> >> >>
> >> >> on
> >> >>
> >> >> > those settings.
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
> >> >>ers http://www.freeswitch.org
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> >> >rs http://www.freeswitch.org
> >>
> >> On Fri, May 21, 2010 at 4:34 AM, Carlos Romero
> >> <romerocarlos24 at gmail.com>
> >
> > wrote:
> >> > what do you mean that you don't have access to it?
> >> >
> >> > you have the code
> >> >
> >> > On Thu, May 20, 2010 at 3:23 PM, Anthony Minessale
> >> >
> >> > <anthony.minessale at gmail.com> wrote:
> >> >> not without intervention from the sofia devs,
> >> >> Sofia sip does that deep down and we don't have access to it.
> >> >> even when we supply it as a one big string it renders it out
> >> >> depending on those settings.
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> 



From steveu at coppice.org  Sat May 22 18:59:37 2010
From: steveu at coppice.org (Steve Underwood)
Date: Sun, 23 May 2010 09:59:37 +0800
Subject: [Freeswitch-users] mod_dptools/redirect and 300 Multiple Choices
In-Reply-To: <AANLkTinJ04EHUyvEw8sv81DC3HR7Chd7FywxZVpVhkP6@mail.gmail.com>
References: <AANLkTinYtV2CL1H_hME51kP4OpJLaYeuQK0jUEnrimAz@mail.gmail.com>	<AANLkTikfqwv_3UxrKOqQTKaE518ENhA5zw7Usoj3vH_G@mail.gmail.com>	<AANLkTikmqhQN5lRLKUZdvq-78gFEOoQpihLxx0yOsnh6@mail.gmail.com>	<201005221553.42315.sos@sokhapkin.dyndns.org>
	<AANLkTinJ04EHUyvEw8sv81DC3HR7Chd7FywxZVpVhkP6@mail.gmail.com>
Message-ID: <4BF88C09.80503@coppice.org>

On 05/23/2010 08:27 AM, Carlos Romero wrote:
> http://www.pjsip.org/
>    
PJSIP as a substitute for Sofia in high volume server applications? Are 
you a jester or a troll? :-)

All stacks have limitations, and Sofia has some. In the early days of FS 
development, every open source SIP stack was tried. All the alternatives 
were found to have far more limitations than Sofia.
> On Sat, May 22, 2010 at 3:53 PM, Sergey Okhapkin
> <sos at sokhapkin.dyndns.org>  wrote:
>    
>> Could you suggest a better open source SIP implementation?
>>
>> On Friday 21 May 2010, Carlos Romero wrote:
>>      
>>> why choose  a library that is hard to work with in the first place?
>>>
>>> it seems like sofia has many downsides... limited threads per
>>> sip-profile and now this
>>>
>>> On Fri, May 21, 2010 at 3:20 PM, Michael Collins<msc at freeswitch.org>
>>>        
>> wrote:
>>      
>>>> On Thu, May 20, 2010 at 11:46 PM, Carlos Romero<romerocarlos24 at
>>>>          
>> gmail.com>wrote:
>>      
>>>>> what do you mean that you don't have access to it?
>>>>>
>>>>> you have the code
>>>>>            
>>>> The SIP stack is supplied by Nokia. Open source or not, it is bad form to
>>>> muck with someone else's library without due cause. Like all robust SIP
>>>> stacks, Sofia is huge, deep, complex, and otherwise uninviting of those
>>>> who would waltz in and do whatever they wish. If you feel that it is even
>>>> within the realm of possibility then I recommend that you find the
>>>> location in the Sofia library where this takes place, make the
>>>> appropriate changes, and thoroughly test for regressions. Submit the
>>>> patch to the Sofia devs who will then evaluate it and see if it is worthy
>>>> of inclusion.
>>>>
>>>> Or more simply put: "having the code" and "having access" are COMPLETELY
>>>> different concepts.
>>>> -MC
>>>>
>>>>          
>>>>> On Thu, May 20, 2010 at 3:23 PM, Anthony Minessale
>>>>>
>>>>> <anthony.minessale at gmail.com>  wrote:
>>>>>            
>>>>>> not without intervention from the sofia devs,
>>>>>> Sofia sip does that deep down and we don't have access to it.
>>>>>> even when we supply it as a one big string it renders it out depending
>>>>>>              
>>>>> on
>>>>>
>>>>>            
>>>>>> those settings.
>>>>>>              
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>            
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>          
>>> On Fri, May 21, 2010 at 4:34 AM, Carlos Romero<romerocarlos24 at gmail.com>
>>>        
>> wrote:
>>      
>>>> what do you mean that you don't have access to it?
>>>>
>>>> you have the code
>>>>
>>>> On Thu, May 20, 2010 at 3:23 PM, Anthony Minessale
>>>>
>>>> <anthony.minessale at gmail.com>  wrote:
>>>>          
>>>>> not without intervention from the sofia devs,
>>>>> Sofia sip does that deep down and we don't have access to it.
>>>>> even when we supply it as a one big string it renders it out depending
>>>>> on those settings.
>>>>>            
>>>
Steve



From babak.freeswitch at gmail.com  Sat May 22 21:39:13 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Sun, 23 May 2010 09:09:13 +0430
Subject: [Freeswitch-users] mod_managed session.recordFile() problem
Message-ID: <AANLkTine15DGTikyP_ycHNrJWwEqIPukirUiS-oYx_Rd@mail.gmail.com>

Hi all
somebody save me plz! I'm using the code below to simply record messages:
Session.Answer()
...
if (Session.Ready())
            {

                Session.Execute("sleep", "100");
                Session.StreamFile(..., -1);
                Session.Execute("playback",
"tone_stream://%(500,500,480,620)");
                Session.RecordFile(..., 60000, 500, 5);
            }
it's working on my pc with no problem but when I upload it to the
server(which is much much much stronger than my pc) first 5 6 sec of message
is not recorded
thanks
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From talk2ram at gmail.com  Sat May 22 23:09:47 2010
From: talk2ram at gmail.com (ram)
Date: Sun, 23 May 2010 11:39:47 +0530
Subject: [Freeswitch-users] Gateway Registration Issues
In-Reply-To: <4F8EA7B9-B642-4C66-AF5A-ADDF554DA166@gmail.com>
References: <AANLkTim-d2_nD3lLS3YlqciwrgJpGP5jMyCfSWtxI_Ly@mail.gmail.com>
	<983B7F89-A86E-4AFE-91E6-553D1F0B6495@gmail.com>
	<AANLkTil3UJKkxDU1XHt6FNIpfccng5ynLSnhjMq2kgq_@mail.gmail.com>
	<4F8EA7B9-B642-4C66-AF5A-ADDF554DA166@gmail.com>
Message-ID: <AANLkTik0NEOMtkpTbswnxQu3p6c8-4G8cW32aEfgaiLC@mail.gmail.com>

Hi

yes i have stopped the deamon and started again
but i see still it show the same

On Fri, May 21, 2010 at 12:35 PM, David Ponzone <david.ponzone at gmail.com>wrote:

> You restarted the gateway ?
>
> sofia profile external killgw mydomain
> sofia profile external rescan reloadxml
>
>   David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
>  tel:      0811 46 26 26
>  www.ipeva.fr  -   www.ipeva-studio.com
>
>  *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
>  Le 21/05/2010 ? 07:00, ram a ?crit :
>
>
>
>
> thats good suggstion
>
> iam working on same problem
>
> <gateway name="mydomain">
>       <param name="username" value="2000"/>
>       <param name="auth-username" value="2000"/>
>       <param name="password" value="XXXX"/>
>       <param name="from-user" value="2000"/>
>       <param name="from-domain" value="sip5.domain.com"/>
>       <param name="proxy" value="sip5.domain.com"/>
>       <param name="register-proxy" value="sip5.domain.com"/>
>       <param name="outbound-proxy" value="sip5.domain.com"/>
>       <param name="expire-seconds" value="800"/>
>       <param name="register" value="true"/>
>       <param name="register-transport" value="udp"/>
>       <param name="retry-seconds" value="60"/>
>       <param name="ping" value="6000"/>
>       <param name="context" value="public"/>
>       <param name="caller-id-in-from" value="true"/>
>       <param name="extension_in_contact" value="true"/>
>     </gateway>
> </include>
>
>
> and My gateway side still show like below
>
>
>
> AOR:: 2000 at sip5.otherdomain.com
>                 Contact::
> sip:gw+mydomain at x.x.x.x:5080;transport=udp;gw=mydomain Q=
>                         Expires:: 794
>                         Callid:: 30b6aa56-6491-11df-86f8-797747db92ac
>                         Cseq:: 131089695
>                         User-agent::
> FreeSWITCH-mod_sofia/1.0.head-svn-17503
>                         State:: CS_NEW
>                         Flags:: 0
>                         Cflag:: 0
>                         Socket:: udp:x.x.x.x:5060
>                         Methods:: 5631
>
>
> On Thu, May 13, 2010 at 6:28 PM, David Ponzone <david.ponzone at gmail.com>wrote:
>
>> Aloysius,,
>>
>> That is an old-skool carrier.
>> You have to add this line to your gateway params:
>>
>> <param name="extension-in-contact" value="true"/>
>>
>>
>>   David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>>  tel:      0811 46 26 26
>>  www.ipeva.fr  -   www.ipeva-studio.com
>>
>>  *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>>  Le 13/05/2010 ? 14:46, Aloysius Lloyd a ?crit :
>>
>>   Hi All,
>>
>> I am trying to connect IRISTEL ( CLEC ) in Canada and FreeSWITCH. I have
>> the following gateway configuration
>>
>>  <include>
>>   <gateway name="wwc_test">
>>     <param name="auth-username" value="username"/>
>>     <param name="username" value="14161231234" />
>>     <param name="password" value="password" />
>>     <param name="proxy" value="sbc1.iristel.net" />
>>     <param name="realm" value="iristel.net" />
>>     <param name="expire-seconds" value="120"/>
>>     <param name="register" value="true"/>
>>     <param name="retry-seconds" value="30"/>
>>     <param name="ping" value="25"/>
>>   </gateway>
>> </include>
>>
>>
>> The above configuration is not working. I have the following error in cli
>> Error.
>>
>>  *2010-05-13 08:34:15.987465 [ERR] sofia_reg.c:1501 14161231234
>> Registration Failed with status Conflict [409]. failure #1*
>>
>> ====
>>
>> Also carrier saying there is conflict in the contact field.Here is how
>> they receive now.
>>
>>  *Contact: <
>> sip:gw+test_iristel at 10.20.30.1:5080;transport=udp;gw=test_iristel>.*
>>
>> But the contact field should be like below
>>
>> Contact: <14161231234 at 10.20.30.1:5080;
>>
>> Any suggestions?
>>
>> Thanks in advance.
>>
>> Lloyd
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From stevendt at primrosebank.net  Sun May 23 05:17:12 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Sun, 23 May 2010 13:17:12 +0100
Subject: [Freeswitch-users] Mod_skinny: Call for testing!
References: <AANLkTik8hHRtOci8qTXmZVHOWvVxRg-OPm5zyRAMrWEe@mail.gmail.com><F32D0E59CC994113BC88C7902AFC87BF@bp1.ad.bp.com>,
	<AANLkTikJ1biYWWY-M4qYMWgsYfGelxvFu91EVnBqDlcq@mail.gmail.com><549CFEF87AEDE841A38E9D15EAB4C04C557DE57F78@cooper><3A8B4A19CB3C4270B20D9980B9E02F45@bp1.ad.bp.com>
	<D7473B43-1129-4FDD-B956-D6430C562C64@jerris.com>
Message-ID: <31FDEB4CBC114E3D880E926599C94A5B@bp1.ad.bp.com>

Thanks Mike - it's there!

regards
Dave

----- Original Message ----- 
From: "Michael Jerris" <mike at jerris.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, May 22, 2010 11:25 PM
Subject: Re: [Freeswitch-users] Mod_skinny: Call for testing!


> It should be updated now.
>
> Mike
>
> On May 21, 2010, at 12:52 PM, Dave Stevenson wrote:
>
>> Hi,
>>
>> thanks Peter.
>>
>> I did not realise that the Windows build file was so new, OK, I guess 
>> that
>> I'll just have to wait for the SVN to catch up. I don't really want to
>> update to Git at the moment,
>>
>> regards
>> Dave
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From stevendt at primrosebank.net  Sun May 23 05:22:22 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Sun, 23 May 2010 13:22:22 +0100
Subject: [Freeswitch-users] Mod_skinny: Call for testing!
References: <AANLkTik8hHRtOci8qTXmZVHOWvVxRg-OPm5zyRAMrWEe@mail.gmail.com><F32D0E59CC994113BC88C7902AFC87BF@bp1.ad.bp.com>,
	<AANLkTikJ1biYWWY-M4qYMWgsYfGelxvFu91EVnBqDlcq@mail.gmail.com><549CFEF87AEDE841A38E9D15EAB4C04C557DE57F78@cooper>,
	<3FDDF9772BE64AB8A81243DD375E46BC@bp1.ad.bp.com><549CFEF87AEDE841A38E9D15EAB4C04C557DE57F86@cooper>,
	<CB226C919BE944B59B0934AB1C644169@bp1.ad.bp.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C557DE57F87@cooper>
Message-ID: <DCA63AC1BACA4084A92017299BBC13D8@bp1.ad.bp.com>

Thanks Peter,

MJ has updated the Git, so I have it now.

I've looked at the 2008 Solution and 2008.express Solution and it looks like 
I can just cut and paste the mod_skinny project from the 2008 into the 
express solution. I tried it with WordPad and Visual Studio detected that it 
was an unknown version number but it seemed to work fine when I used 
Notepad.

I have built the solution and it all seemed to go OK, I have not tried to do 
it on the running FreeSwitch server yet - just wanted to check that what I'd 
done was an acceptable way of making the change - what do you think ?

regards
Dave


----- Original Message ----- 
From: "Peter Olsson" <peter.olsson at visionutveckling.se>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, May 22, 2010 11:19 PM
Subject: Re: [Freeswitch-users] Mod_skinny: Call for testing!


Dave,

It seems like it was never added, I've asked Mathieu to add this file as 
well. I never noticed, since I already had my own local copy... If you want 
to, it's possible to download it directly from this link: 
http://jira.freeswitch.org/browse/MODSKINNY-3.

/Peter

________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org 
[freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Dave Stevenson 
[stevendt at primrosebank.net]
Skickat: den 22 maj 2010 23:01
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Mod_skinny: Call for testing!

Hi Peter,

thanks a lot - I'd appreciate it if you can take a look when you get time -
no great hurry. I have a couple of sccp phones that I'd like to get working
but there's no real urgency.

There's something quite odd going on though - well, I don't understand it !

Just looked in the VS2008 Solution file and see a reference to
mod_skinny.vcproj - there's no matching file in my git download ?

regards
Dave



----- Original Message -----
From: "Peter Olsson" <peter.olsson at visionutveckling.se>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, May 22, 2010 7:14 PM
Subject: Re: [Freeswitch-users] Mod_skinny: Call for testing!


Hi Dave,

Actually, the patch was just made for Visual Studio, not the Express
version. But I think you should be able to figure it out by comparing
Freeswitch.2008.sln to Freeswitch.2008.express.sln.

I will try to make some time to get it fixed for express version during the
next week.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org
[freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Dave Stevenson
[stevendt at primrosebank.net]
Skickat: den 22 maj 2010 20:05
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Mod_skinny: Call for testing!

Peter,

after some hiccups, I now have Git installed and just downloaded the latest
Git (21/05/10).

I don't see modskinny in the Visual Studio 2008 Express Solution - how are
you building it ?

regards
Dave


----- Original Message -----
From: "Peter Olsson" <peter.olsson at visionutveckling.se>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Friday, May 21, 2010 4:16 PM
Subject: Re: [Freeswitch-users] Mod_skinny: Call for testing!


I tried building today, and it builds on Windows on latest git - just needs
some testing now :)

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org
[freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Mathieu Parent
[math.parent at gmail.com]
Skickat: den 21 maj 2010 16:24
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Mod_skinny: Call for testing!

On Fri, May 21, 2010 at 3:46 PM, Dave Stevenson
<stevendt at primrosebank.net> wrote:
> Mathieu,
>
> just updated to 17539 (SVN) and skinny is not included in the Windows
> Build
> Solution to VS2008
>
> Does it need to be compiled & linked outside of the VS Build ?

SVN is out of sync, git has Windows support since yesterday:

commit bfb6a8b1f5f835008d1e0fbd90d2e120814f341a
Author: Mathieu Parent <math.parent at gmail.com>
Date:   Thu May 20 23:19:10 2010 +0200

    Skinny: Initial Windows support

    Thanks to Peter Olsson

    Closes: MODSKINNY-3

I have not been able to test it, you can comment MODSKINNY-3 if you
still have problem after the patch applied.

Mathieu Parent

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




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!DSPAM:4bf847d432933212913465!


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From peter.olsson at visionutveckling.se  Sun May 23 05:34:49 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sun, 23 May 2010 14:34:49 +0200
Subject: [Freeswitch-users] Mod_skinny: Call for testing!
In-Reply-To: <DCA63AC1BACA4084A92017299BBC13D8@bp1.ad.bp.com>
References: <AANLkTik8hHRtOci8qTXmZVHOWvVxRg-OPm5zyRAMrWEe@mail.gmail.com><F32D0E59CC994113BC88C7902AFC87BF@bp1.ad.bp.com>,
	<AANLkTikJ1biYWWY-M4qYMWgsYfGelxvFu91EVnBqDlcq@mail.gmail.com><549CFEF87AEDE841A38E9D15EAB4C04C557DE57F78@cooper>,
	<3FDDF9772BE64AB8A81243DD375E46BC@bp1.ad.bp.com><549CFEF87AEDE841A38E9D15EAB4C04C557DE57F86@cooper>,
	<CB226C919BE944B59B0934AB1C644169@bp1.ad.bp.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C557DE57F87@cooper>,
	<DCA63AC1BACA4084A92017299BBC13D8@bp1.ad.bp.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C557DE57F8C@cooper>

Yes, it's probably ok, and if you got it built it must have been :)

I just made the patch to make it build on Windows, all I ever tried to do was to load the module, and it did load.

But I don't have any Cisco phones, so I couldn't try it out more then this :)

/Peter

________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Dave Stevenson [stevendt at primrosebank.net]
Skickat: den 23 maj 2010 14:22
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Mod_skinny: Call for testing!

Thanks Peter,

MJ has updated the Git, so I have it now.

I've looked at the 2008 Solution and 2008.express Solution and it looks like
I can just cut and paste the mod_skinny project from the 2008 into the
express solution. I tried it with WordPad and Visual Studio detected that it
was an unknown version number but it seemed to work fine when I used
Notepad.

I have built the solution and it all seemed to go OK, I have not tried to do
it on the running FreeSwitch server yet - just wanted to check that what I'd
done was an acceptable way of making the change - what do you think ?

regards
Dave


----- Original Message -----
From: "Peter Olsson" <peter.olsson at visionutveckling.se>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, May 22, 2010 11:19 PM
Subject: Re: [Freeswitch-users] Mod_skinny: Call for testing!


Dave,

It seems like it was never added, I've asked Mathieu to add this file as
well. I never noticed, since I already had my own local copy... If you want
to, it's possible to download it directly from this link:
http://jira.freeswitch.org/browse/MODSKINNY-3.

/Peter

________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org
[freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Dave Stevenson
[stevendt at primrosebank.net]
Skickat: den 22 maj 2010 23:01
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Mod_skinny: Call for testing!

Hi Peter,

thanks a lot - I'd appreciate it if you can take a look when you get time -
no great hurry. I have a couple of sccp phones that I'd like to get working
but there's no real urgency.

There's something quite odd going on though - well, I don't understand it !

Just looked in the VS2008 Solution file and see a reference to
mod_skinny.vcproj - there's no matching file in my git download ?

regards
Dave



----- Original Message -----
From: "Peter Olsson" <peter.olsson at visionutveckling.se>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, May 22, 2010 7:14 PM
Subject: Re: [Freeswitch-users] Mod_skinny: Call for testing!


Hi Dave,

Actually, the patch was just made for Visual Studio, not the Express
version. But I think you should be able to figure it out by comparing
Freeswitch.2008.sln to Freeswitch.2008.express.sln.

I will try to make some time to get it fixed for express version during the
next week.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org
[freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Dave Stevenson
[stevendt at primrosebank.net]
Skickat: den 22 maj 2010 20:05
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Mod_skinny: Call for testing!

Peter,

after some hiccups, I now have Git installed and just downloaded the latest
Git (21/05/10).

I don't see modskinny in the Visual Studio 2008 Express Solution - how are
you building it ?

regards
Dave


----- Original Message -----
From: "Peter Olsson" <peter.olsson at visionutveckling.se>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Friday, May 21, 2010 4:16 PM
Subject: Re: [Freeswitch-users] Mod_skinny: Call for testing!


I tried building today, and it builds on Windows on latest git - just needs
some testing now :)

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org
[freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Mathieu Parent
[math.parent at gmail.com]
Skickat: den 21 maj 2010 16:24
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Mod_skinny: Call for testing!

On Fri, May 21, 2010 at 3:46 PM, Dave Stevenson
<stevendt at primrosebank.net> wrote:
> Mathieu,
>
> just updated to 17539 (SVN) and skinny is not included in the Windows
> Build
> Solution to VS2008
>
> Does it need to be compiled & linked outside of the VS Build ?

SVN is out of sync, git has Windows support since yesterday:

commit bfb6a8b1f5f835008d1e0fbd90d2e120814f341a
Author: Mathieu Parent <math.parent at gmail.com>
Date:   Thu May 20 23:19:10 2010 +0200

    Skinny: Initial Windows support

    Thanks to Peter Olsson

    Closes: MODSKINNY-3

I have not been able to test it, you can comment MODSKINNY-3 if you
still have problem after the patch applied.

Mathieu Parent

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




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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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!DSPAM:4bf91f6532932835319569!



From stevendt at primrosebank.net  Sun May 23 09:35:19 2010
From: stevendt at primrosebank.net (Dave Stevenson)
Date: Sun, 23 May 2010 17:35:19 +0100
Subject: [Freeswitch-users] Mod_skinny: Call for testing!
References: <AANLkTik8hHRtOci8qTXmZVHOWvVxRg-OPm5zyRAMrWEe@mail.gmail.com><F32D0E59CC994113BC88C7902AFC87BF@bp1.ad.bp.com>,
	<AANLkTikJ1biYWWY-M4qYMWgsYfGelxvFu91EVnBqDlcq@mail.gmail.com><549CFEF87AEDE841A38E9D15EAB4C04C557DE57F78@cooper>,
	<3FDDF9772BE64AB8A81243DD375E46BC@bp1.ad.bp.com><549CFEF87AEDE841A38E9D15EAB4C04C557DE57F86@cooper>,
	<CB226C919BE944B59B0934AB1C644169@bp1.ad.bp.com><549CFEF87AEDE841A38E9D15EAB4C04C557DE57F87@cooper>,
	<DCA63AC1BACA4084A92017299BBC13D8@bp1.ad.bp.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C557DE57F8C@cooper>
Message-ID: <ABFFF2C456EC47398B0A43688B67245A@bp1.ad.bp.com>

Thanks Peter,

I've run out of time today, but I will try updating the running server next 
week and see if it works with a couple of Cisco phones - I'll keep you 
posted !

regards
Dave


----- Original Message ----- 
From: "Peter Olsson" <peter.olsson at visionutveckling.se>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Sunday, May 23, 2010 1:34 PM
Subject: Re: [Freeswitch-users] Mod_skinny: Call for testing!


Yes, it's probably ok, and if you got it built it must have been :)

I just made the patch to make it build on Windows, all I ever tried to do 
was to load the module, and it did load.

But I don't have any Cisco phones, so I couldn't try it out more then this 
:)

/Peter

________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org 
[freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Dave Stevenson 
[stevendt at primrosebank.net]
Skickat: den 23 maj 2010 14:22
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Mod_skinny: Call for testing!

Thanks Peter,

MJ has updated the Git, so I have it now.

I've looked at the 2008 Solution and 2008.express Solution and it looks like
I can just cut and paste the mod_skinny project from the 2008 into the
express solution. I tried it with WordPad and Visual Studio detected that it
was an unknown version number but it seemed to work fine when I used
Notepad.

I have built the solution and it all seemed to go OK, I have not tried to do
it on the running FreeSwitch server yet - just wanted to check that what I'd
done was an acceptable way of making the change - what do you think ?

regards
Dave


----- Original Message -----
From: "Peter Olsson" <peter.olsson at visionutveckling.se>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, May 22, 2010 11:19 PM
Subject: Re: [Freeswitch-users] Mod_skinny: Call for testing!


Dave,

It seems like it was never added, I've asked Mathieu to add this file as
well. I never noticed, since I already had my own local copy... If you want
to, it's possible to download it directly from this link:
http://jira.freeswitch.org/browse/MODSKINNY-3.

/Peter

________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org
[freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Dave Stevenson
[stevendt at primrosebank.net]
Skickat: den 22 maj 2010 23:01
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Mod_skinny: Call for testing!

Hi Peter,

thanks a lot - I'd appreciate it if you can take a look when you get time -
no great hurry. I have a couple of sccp phones that I'd like to get working
but there's no real urgency.

There's something quite odd going on though - well, I don't understand it !

Just looked in the VS2008 Solution file and see a reference to
mod_skinny.vcproj - there's no matching file in my git download ?

regards
Dave



----- Original Message -----
From: "Peter Olsson" <peter.olsson at visionutveckling.se>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Saturday, May 22, 2010 7:14 PM
Subject: Re: [Freeswitch-users] Mod_skinny: Call for testing!


Hi Dave,

Actually, the patch was just made for Visual Studio, not the Express
version. But I think you should be able to figure it out by comparing
Freeswitch.2008.sln to Freeswitch.2008.express.sln.

I will try to make some time to get it fixed for express version during the
next week.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org
[freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Dave Stevenson
[stevendt at primrosebank.net]
Skickat: den 22 maj 2010 20:05
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Mod_skinny: Call for testing!

Peter,

after some hiccups, I now have Git installed and just downloaded the latest
Git (21/05/10).

I don't see modskinny in the Visual Studio 2008 Express Solution - how are
you building it ?

regards
Dave


----- Original Message -----
From: "Peter Olsson" <peter.olsson at visionutveckling.se>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Friday, May 21, 2010 4:16 PM
Subject: Re: [Freeswitch-users] Mod_skinny: Call for testing!


I tried building today, and it builds on Windows on latest git - just needs
some testing now :)

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org
[freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Mathieu Parent
[math.parent at gmail.com]
Skickat: den 21 maj 2010 16:24
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Mod_skinny: Call for testing!

On Fri, May 21, 2010 at 3:46 PM, Dave Stevenson
<stevendt at primrosebank.net> wrote:
> Mathieu,
>
> just updated to 17539 (SVN) and skinny is not included in the Windows
> Build
> Solution to VS2008
>
> Does it need to be compiled & linked outside of the VS Build ?

SVN is out of sync, git has Windows support since yesterday:

commit bfb6a8b1f5f835008d1e0fbd90d2e120814f341a
Author: Mathieu Parent <math.parent at gmail.com>
Date:   Thu May 20 23:19:10 2010 +0200

    Skinny: Initial Windows support

    Thanks to Peter Olsson

    Closes: MODSKINNY-3

I have not been able to test it, you can comment MODSKINNY-3 if you
still have problem after the patch applied.

Mathieu Parent

_______________________________________________
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From davis.erwin at gmail.com  Sun May 23 15:27:44 2010
From: davis.erwin at gmail.com (Erwin Davis)
Date: Sun, 23 May 2010 15:27:44 -0700
Subject: [Freeswitch-users] curl_xml
In-Reply-To: <03E6E212-1AF1-4CB3-8521-FBAD1DA53B7B@freeswitch.org>
References: <AANLkTiltmx-qzAAgkQjUcCAGsAMp9_X573zbQic24pf7@mail.gmail.com>
	<DF566DB4-AA3D-468C-A746-93F57BFED1D3@freeswitch.org>
	<AANLkTimFLCMqXYm27dq9oHPVtJ3pdldblhXcaOGu31BZ@mail.gmail.com>
	<03E6E212-1AF1-4CB3-8521-FBAD1DA53B7B@freeswitch.org>
Message-ID: <AANLkTinDXvb_-69tK55KZIVurLjSkEP5cMrMTnXs7DXc@mail.gmail.com>

Hi, Brian,

I carefully follow up the template from the link you provided, but my x-lit
client  still received 403 forbidden response. What should I look at next?
Below is the xml response from my web server. Thanks,

<document type="freeswitch/xml">
             <section name="directory" description="User Directory">
                 <domain name="192.168.1.31">
                       <groups>
                           <group name="default">
                               <users>
                                  <user id="1018">
                                     <params>
                                        <param name="password"
value="1018"/>
                                     </params>
                                  </user>
                               </users>
                           </group>
                         </groups>
                   </domain>
             </section>
         </document>


e
On Wed, May 19, 2010 at 7:42 AM, Brian West <brian at freeswitch.org> wrote:

> Yes that is wrong you're missing some stuff please read:
> http://wiki.freeswitch.org/wiki/Xml_curl#Response
>
> /b
>
> On May 19, 2010, at 9:40 AM, Erwin Davis wrote:
>
> the xml log file is in http://pastebin.freeswitch.org/13006<http://pastebin.freeswitch.org/13007>
>
> The webserver sent back the response as below,
>
>    1.   <domain name="192.168.1.31">
>    2.                     <user id="1018">
>    3.                         <params>
>    4.                             <param name="password" value="1018"/>
>    5.                         </params>
>    6.                     </user>
>    7.                 </domain>
>    8.
>
>
>
>
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From bobc at devassert.com  Sun May 23 16:08:08 2010
From: bobc at devassert.com (Bob Coleman)
Date: Mon, 24 May 2010 11:08:08 +1200
Subject: [Freeswitch-users] How do you set outbound from address in SIP
	header
Message-ID: <AANLkTimED3tWvvF9XeecAe298QTQWj8jy_9WlOoO7pqi@mail.gmail.com>

Hi,

I am trying to set the sip from address for outbound calls, but cant
seem to find the option.

The below is from a sip trace, and shows it set to the defaults
eg From: "FreeSWITCH" <sip:0000000000 at 10.12.78.1>;tag=pQ68N9mKc7rDK

I have changed the vars.xml to set the above details but nothing
happens, restarted just to make sure. Searched the other conf files to
make sure there is not another place.

Using an older binary 1.4 , is there any known issues regarding that.

Thanks

Bob


From bobc at devassert.com  Sun May 23 17:38:50 2010
From: bobc at devassert.com (Bob Coleman)
Date: Mon, 24 May 2010 12:38:50 +1200
Subject: [Freeswitch-users] How do you set outbound from address in SIP
	header
In-Reply-To: <AANLkTimED3tWvvF9XeecAe298QTQWj8jy_9WlOoO7pqi@mail.gmail.com>
References: <AANLkTimED3tWvvF9XeecAe298QTQWj8jy_9WlOoO7pqi@mail.gmail.com>
Message-ID: <AANLkTin2oG76xIHYfARa9b5Lr0ppRqDeykVsN9lX1GRR@mail.gmail.com>

Never mind.

The sip from header is set by:
origination_caller_id_number and _name

when using originate

Thanks

On Mon, May 24, 2010 at 11:08 AM, Bob Coleman <bobc at devassert.com> wrote:
> Hi,
>
> I am trying to set the sip from address for outbound calls, but cant
> seem to find the option.
>
> The below is from a sip trace, and shows it set to the defaults
> eg From: "FreeSWITCH" <sip:0000000000 at 10.12.78.1>;tag=pQ68N9mKc7rDK
>
> I have changed the vars.xml to set the above details but nothing
> happens, restarted just to make sure. Searched the other conf files to
> make sure there is not another place.
>
> Using an older binary 1.4 , is there any known issues regarding that.
>
> Thanks
>
> Bob
>


From daniel.neubert at solomo.de  Sun May 23 17:40:28 2010
From: daniel.neubert at solomo.de (Daniel Neubert)
Date: Mon, 24 May 2010 02:40:28 +0200
Subject: [Freeswitch-users] curl_xml
In-Reply-To: <AANLkTinDXvb_-69tK55KZIVurLjSkEP5cMrMTnXs7DXc@mail.gmail.com>
References: <AANLkTiltmx-qzAAgkQjUcCAGsAMp9_X573zbQic24pf7@mail.gmail.com>	<DF566DB4-AA3D-468C-A746-93F57BFED1D3@freeswitch.org>	<AANLkTimFLCMqXYm27dq9oHPVtJ3pdldblhXcaOGu31BZ@mail.gmail.com>	<03E6E212-1AF1-4CB3-8521-FBAD1DA53B7B@freeswitch.org>
	<AANLkTinDXvb_-69tK55KZIVurLjSkEP5cMrMTnXs7DXc@mail.gmail.com>
Message-ID: <4BF9CAFC.7090302@solomo.de>

Hi Erwin,

Your XML does not look valid/complete - please refer to 
http://wiki.freeswitch.org/wiki/Xml_curl#Authorization for an example 
response.

My XML looks like this (stripped down to minimal requirements):

<document type="freeswitch/xml">
<section name="directory" description="User Profile">
<domain name="{domain}">
<user id="USERNAME">
<params>
<param name="password" value="PASSWORD"/>
</params>
<variables>
<variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/>
</variables>
</user>
</domain>
</section>
</document>

Regards,
Daniel

On 24.05.2010 00:27, Erwin Davis wrote:
> Hi, Brian,
>
> I carefully follow up the template from the link you provided, but my 
> x-lit client  still received 403 forbidden response. What should I 
> look at next?
> Below is the xml response from my web server. Thanks,
>
> <document type="freeswitch/xml">
> <section name="directory" description="User Directory">
> <domain name="192.168.1.31">
> <groups>
> <group name="default">
> <users>
> <user id="1018">
> <params>
> <param name="password" value="1018"/>
> </params>
> </user>
> </users>
> </group>
> </groups>
> </domain>
> </section>
> </document>
>
>
> e
> On Wed, May 19, 2010 at 7:42 AM, Brian West <brian at freeswitch.org 
> <mailto:brian at freeswitch.org>> wrote:
>
>     Yes that is wrong you're missing some stuff please read:
>     http://wiki.freeswitch.org/wiki/Xml_curl#Response
>
>     /b
>
>     On May 19, 2010, at 9:40 AM, Erwin Davis wrote:
>
>>     the xml log file is in http://pastebin.freeswitch.org/13006
>>     <http://pastebin.freeswitch.org/13007>
>>
>>     The webserver sent back the response as below,
>>
>>       1.
>>           <domain name="192.168.1.31">
>>       2.
>>           <user id="1018">
>>       3.
>>           <params>
>>       4.
>>           <param name="password" value="1018"/>
>>       5.
>>           </params>
>>       6.
>>           </user>
>>       7.
>>           </domain>
>>       8.
>>
>>
>
>
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From davis.erwin at gmail.com  Sun May 23 18:45:56 2010
From: davis.erwin at gmail.com (Erwin Davis)
Date: Sun, 23 May 2010 18:45:56 -0700
Subject: [Freeswitch-users] curl_xml
In-Reply-To: <4BF9CAFC.7090302@solomo.de>
References: <AANLkTiltmx-qzAAgkQjUcCAGsAMp9_X573zbQic24pf7@mail.gmail.com>
	<DF566DB4-AA3D-468C-A746-93F57BFED1D3@freeswitch.org>
	<AANLkTimFLCMqXYm27dq9oHPVtJ3pdldblhXcaOGu31BZ@mail.gmail.com>
	<03E6E212-1AF1-4CB3-8521-FBAD1DA53B7B@freeswitch.org>
	<AANLkTinDXvb_-69tK55KZIVurLjSkEP5cMrMTnXs7DXc@mail.gmail.com>
	<4BF9CAFC.7090302@solomo.de>
Message-ID: <AANLkTin2KFDpjFdxDIYsocYg1bH0bIQsExKnrrOtHM73@mail.gmail.com>

Hi, Daniel,

I modified my xml template to be same as yours, but still not work. Could it
be something else wrong? How to debug? Appreciate if any tips. Thanks,

<document type="freeswitch/xml">
             <section name="directory" description="User Directory">
                 <domain name="192.168.1.31">
                      <user id="1018">
                          <params>
                              <param name="password" value="1018"/>
                          </params>
                          <variables>
                              <variable name="sip-force-contact"
value="NDLB-connectile-dysfunction"/>
                          </variables>
                      </user>
                 </domain>
             </section>
         </document>

On Sun, May 23, 2010 at 5:40 PM, Daniel Neubert <daniel.neubert at solomo.de>wrote:

>  Hi Erwin,
>
> Your XML does not look valid/complete - please refer to
> http://wiki.freeswitch.org/wiki/Xml_curl#Authorization for an example
> response.
>
> My XML looks like this (stripped down to minimal requirements):
>
> <document type="freeswitch/xml">
>     <section name="directory" description="User Profile">
>         <domain name="{domain}">
>             <user id="USERNAME">
>                 <params>
>                     <param name="password" value="PASSWORD"/>
>                 </params>
>                 <variables>
>                     <variable name="sip-force-contact"
> value="NDLB-connectile-dysfunction"/>
>                 </variables>
>
>             </user>
>         </domain>
>     </section>
> </document>
>
> Regards,
> Daniel
>
> On 24.05.2010 00:27, Erwin Davis wrote:
>
> Hi, Brian,
>
> I carefully follow up the template from the link you provided, but my x-lit
> client  still received 403 forbidden response. What should I look at next?
> Below is the xml response from my web server. Thanks,
>
> <document type="freeswitch/xml">
>              <section name="directory" description="User Directory">
>                  <domain name="192.168.1.31">
>                        <groups>
>                            <group name="default">
>                                <users>
>                                   <user id="1018">
>                                      <params>
>                                         <param name="password"
> value="1018"/>
>                                      </params>
>                                   </user>
>                                </users>
>                            </group>
>                          </groups>
>                    </domain>
>              </section>
>          </document>
>
>
> e
> On Wed, May 19, 2010 at 7:42 AM, Brian West <brian at freeswitch.org> wrote:
>
>> Yes that is wrong you're missing some stuff please read:
>> http://wiki.freeswitch.org/wiki/Xml_curl#Response
>>
>>  /b
>>
>>  On May 19, 2010, at 9:40 AM, Erwin Davis wrote:
>>
>> the xml log file is in http://pastebin.freeswitch.org/13006<http://pastebin.freeswitch.org/13007>
>>
>> The webserver sent back the response as below,
>>
>>    1.   <domain name="192.168.1.31">
>>     2.                     <user id="1018">
>>     3.                         <params>
>>     4.                             <param name="password" value="1018"/>
>>     5.                         </params>
>>     6.                     </user>
>>     7.                 </domain>
>>     8.
>>
>>
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Sun May 23 20:43:27 2010
From: msc at freeswitch.org (Michael S Collins)
Date: Sun, 23 May 2010 20:43:27 -0700
Subject: [Freeswitch-users] curl_xml
In-Reply-To: <AANLkTin2KFDpjFdxDIYsocYg1bH0bIQsExKnrrOtHM73@mail.gmail.com>
References: <AANLkTiltmx-qzAAgkQjUcCAGsAMp9_X573zbQic24pf7@mail.gmail.com>
	<DF566DB4-AA3D-468C-A746-93F57BFED1D3@freeswitch.org>
	<AANLkTimFLCMqXYm27dq9oHPVtJ3pdldblhXcaOGu31BZ@mail.gmail.com>
	<03E6E212-1AF1-4CB3-8521-FBAD1DA53B7B@freeswitch.org>
	<AANLkTinDXvb_-69tK55KZIVurLjSkEP5cMrMTnXs7DXc@mail.gmail.com>
	<4BF9CAFC.7090302@solomo.de>
	<AANLkTin2KFDpjFdxDIYsocYg1bH0bIQsExKnrrOtHM73@mail.gmail.com>
Message-ID: <1B2B2257-94F7-47BD-938A-369FB792D2D8@freeswitch.org>

Have you turned in XML curl debugging at the console to see what  
exactly is happening when the XML comes back? Do xml_curl debug on at  
the fscli then make a test call, capturing the console output. Drop it  
in pastebin and link to it in this thread.

-MC

Sent from my iPhone

On May 23, 2010, at 6:45 PM, Erwin Davis <davis.erwin at gmail.com> wrote:

> Hi, Daniel,
>
> I modified my xml template to be same as yours, but still not work.  
> Could it be something else wrong? How to debug? Appreciate if any  
> tips. Thanks,
>
> <document type="freeswitch/xml">
>              <section name="directory" description="User Directory">
>                  <domain name="192.168.1.31">
>                       <user id="1018">
>                           <params>
>                               <param name="password" value="1018"/>
>                           </params>
>                           <variables>
>                               <variable name="sip-force-contact"  
> value="NDLB-connectile-dysfunction"/>
>                           </variables>
>                       </user>
>                  </domain>
>              </section>
>          </document>
>
> On Sun, May 23, 2010 at 5:40 PM, Daniel Neubert <daniel.neubert at solomo.de 
> > wrote:
> Hi Erwin,
>
> Your XML does not look valid/complete - please refer to http://wiki.freeswitch.org/wiki/Xml_curl#Authorization 
>  for an example response.
>
> My XML looks like this (stripped down to minimal requirements):
>
> <document type="freeswitch/xml">
>     <section name="directory" description="User Profile">
>         <domain name="{domain}">
>             <user id="USERNAME">
>                 <params>
>                     <param name="password" value="PASSWORD"/>
>                 </params>
>                 <variables>
>                     <variable name="sip-force-contact" value="NDLB- 
> connectile-dysfunction"/>
>                 </variables>
>             </user>
>         </domain>
>     </section>
> </document>
>
> Regards,
> Daniel
>
> On 24.05.2010 00:27, Erwin Davis wrote:
>>
>> Hi, Brian,
>>
>> I carefully follow up the template from the link you provided, but  
>> my x-lit client  still received 403 forbidden response. What should  
>> I look at next?
>> Below is the xml response from my web server. Thanks,
>>
>> <document type="freeswitch/xml">
>>              <section name="directory" description="User Directory">
>>                  <domain name="192.168.1.31">
>>                        <groups>
>>                            <group name="default">
>>                                <users>
>>                                   <user id="1018">
>>                                      <params>
>>                                         <param name="password"  
>> value="1018"/>
>>                                      </params>
>>                                   </user>
>>                                </users>
>>                            </group>
>>                          </groups>
>>                    </domain>
>>              </section>
>>          </document>
>>
>>
>> e
>> On Wed, May 19, 2010 at 7:42 AM, Brian West <brian at freeswitch.org>  
>> wrote:
>> Yes that is wrong you're missing some stuff please read: http://wiki.freeswitch.org/wiki/Xml_curl#Response
>>
>> /b
>>
>> On May 19, 2010, at 9:40 AM, Erwin Davis wrote:
>>
>>> the xml log file is in http://pastebin.freeswitch.org/13006
>>>
>>> The webserver sent back the response as below,
>>>   <domain name="192.168.1.31">
>>>                     <user id="1018">
>>>                         <params>
>>>                             <param name="password" value="1018"/>
>>>                         </params>
>>>                     </user>
>>>                 </domain>
>>>
>>>
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From romerocarlos24 at gmail.com  Sun May 23 23:03:42 2010
From: romerocarlos24 at gmail.com (Carlos Romero)
Date: Mon, 24 May 2010 02:03:42 -0400
Subject: [Freeswitch-users] mod_dptools/redirect and 300 Multiple Choices
In-Reply-To: <201005222111.30178.sos@sokhapkin.dyndns.org>
References: <AANLkTinYtV2CL1H_hME51kP4OpJLaYeuQK0jUEnrimAz@mail.gmail.com>
	<201005221553.42315.sos@sokhapkin.dyndns.org>
	<AANLkTinJ04EHUyvEw8sv81DC3HR7Chd7FywxZVpVhkP6@mail.gmail.com>
	<201005222111.30178.sos@sokhapkin.dyndns.org>
Message-ID: <AANLkTinHT06-4_CmV5SnaGE01ftZqeZnxl0PRKJ9Kjfs@mail.gmail.com>

Sorry I wasn't aware of this.

So what do you intend to do to remove the limitations of sofia that
seems to affect performance?

Carlos Romero

On Sat, May 22, 2010 at 6:59 PM, Steve Underwood wrote:
> PJSIP as a substitute for Sofia in high volume server applications? Are
> you a jester or a troll? :-)
>
> All stacks have limitations, and Sofia has some. In the early days of FS
> development, every open source SIP stack was tried. All the alternatives
> were found to have far more limitations than Sofia.

On Sat, May 22, 2010 at 9:11 PM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org> wrote:
> http://www.pjsip.org/apps.htm doesn't list any serious server side software
> based on pjsip, but simple client side implementations only. Perhaps there is
> a reason for that...
>
> On Saturday 22 May 2010, Carlos Romero wrote:
>> http://www.pjsip.org/
>>
>> On Sat, May 22, 2010 at 3:53 PM, Sergey Okhapkin
>>
>> <sos at sokhapkin.dyndns.org> wrote:
>> > Could you suggest a better open source SIP implementation?
>> >
>> > On Friday 21 May 2010, Carlos Romero wrote:
>> >> why choose ?a library that is hard to work with in the first place?
>> >>
>> >> it seems like sofia has many downsides... limited threads per
>> >> sip-profile and now this
>> >>
>> >> On Fri, May 21, 2010 at 3:20 PM, Michael Collins <msc at freeswitch.org>
>> >
>> > wrote:
>> >> > On Thu, May 20, 2010 at 11:46 PM, Carlos Romero <romerocarlos24 at
>> >
>> > gmail.com>wrote:
>> >> >> what do you mean that you don't have access to it?
>> >> >>
>> >> >> you have the code
>> >> >
>> >> > The SIP stack is supplied by Nokia. Open source or not, it is bad form
>> >> > to muck with someone else's library without due cause. Like all robust
>> >> > SIP stacks, Sofia is huge, deep, complex, and otherwise uninviting of
>> >> > those who would waltz in and do whatever they wish. If you feel that
>> >> > it is even within the realm of possibility then I recommend that you
>> >> > find the location in the Sofia library where this takes place, make
>> >> > the appropriate changes, and thoroughly test for regressions. Submit
>> >> > the patch to the Sofia devs who will then evaluate it and see if it is
>> >> > worthy of inclusion.
>> >> >
>> >> > Or more simply put: "having the code" and "having access" are
>> >> > COMPLETELY different concepts.
>> >> > -MC
>> >> >
>> >> >> On Thu, May 20, 2010 at 3:23 PM, Anthony Minessale
>> >> >>
>> >> >> <anthony.minessale at gmail.com> wrote:
>> >> >> > not without intervention from the sofia devs,
>> >> >> > Sofia sip does that deep down and we don't have access to it.
>> >> >> > even when we supply it as a one big string it renders it out
>> >> >> > depending
>> >> >>
>> >> >> on
>> >> >>
>> >> >> > those settings.
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
>> >> >>ers http://www.freeswitch.org
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>> >> >rs http://www.freeswitch.org
>> >>
>> >> On Fri, May 21, 2010 at 4:34 AM, Carlos Romero
>> >> <romerocarlos24 at gmail.com>
>> >
>> > wrote:
>> >> > what do you mean that you don't have access to it?
>> >> >
>> >> > you have the code
>> >> >
>> >> > On Thu, May 20, 2010 at 3:23 PM, Anthony Minessale
>> >> >
>> >> > <anthony.minessale at gmail.com> wrote:
>> >> >> not without intervention from the sofia devs,
>> >> >> Sofia sip does that deep down and we don't have access to it.
>> >> >> even when we supply it as a one big string it renders it out
>> >> >> depending on those settings.
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>>
>
>


From mike at jerris.com  Sun May 23 23:22:31 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 24 May 2010 02:22:31 -0400
Subject: [Freeswitch-users] mod_dptools/redirect and 300 Multiple Choices
In-Reply-To: <AANLkTinHT06-4_CmV5SnaGE01ftZqeZnxl0PRKJ9Kjfs@mail.gmail.com>
References: <AANLkTinYtV2CL1H_hME51kP4OpJLaYeuQK0jUEnrimAz@mail.gmail.com>
	<201005221553.42315.sos@sokhapkin.dyndns.org>
	<AANLkTinJ04EHUyvEw8sv81DC3HR7Chd7FywxZVpVhkP6@mail.gmail.com>
	<201005222111.30178.sos@sokhapkin.dyndns.org>
	<AANLkTinHT06-4_CmV5SnaGE01ftZqeZnxl0PRKJ9Kjfs@mail.gmail.com>
Message-ID: <CDBBD80F-B16A-4EF7-B724-A6F2B8A1A678@jerris.com>

wait until several hundred calls per second isn't enough for my needs and then care about it?

This thread officially closed.  If anyone has any actual code to contribute to this, they can do so on http://jira.freeswitch.org.

Mike

On May 24, 2010, at 2:03 AM, Carlos Romero wrote:

> Sorry I wasn't aware of this.
> 
> So what do you intend to do to remove the limitations of sofia that
> seems to affect performance?
> 
> Carlos Romero
> 
> On Sat, May 22, 2010 at 6:59 PM, Steve Underwood wrote:
>> PJSIP as a substitute for Sofia in high volume server applications? Are
>> you a jester or a troll? :-)
>> 
>> All stacks have limitations, and Sofia has some. In the early days of FS
>> development, every open source SIP stack was tried. All the alternatives
>> were found to have far more limitations than Sofia.
> 



From jan.berger at video24.no  Mon May 24 03:42:09 2010
From: jan.berger at video24.no (Jan Berger)
Date: Mon, 24 May 2010 12:42:09 +0200
Subject: [Freeswitch-users] mod_dptools/redirect and 300 Multiple Choices
In-Reply-To: <4BF88C09.80503@coppice.org>
References: <AANLkTinYtV2CL1H_hME51kP4OpJLaYeuQK0jUEnrimAz@mail.gmail.com>
	<AANLkTikfqwv_3UxrKOqQTKaE518ENhA5zw7Usoj3vH_G@mail.gmail.com>
	<AANLkTikmqhQN5lRLKUZdvq-78gFEOoQpihLxx0yOsnh6@mail.gmail.com>
	<201005221553.42315.sos@sokhapkin.dyndns.org>
	<AANLkTinJ04EHUyvEw8sv81DC3HR7Chd7FywxZVpVhkP6@mail.gmail.com>
	<4BF88C09.80503@coppice.org>
Message-ID: <9507FEFEE27A4B9E8D1D85D122A7019F@dell9400>

Don't offend the troll's, they are nice creatures far better than their
reputation :) - even if they never show up when tourists make long walks in
our forests trying to capture a photo of one :)


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steve
Underwood
Sent: 23. mai 2010 04:00
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_dptools/redirect and 300 Multiple
Choices

On 05/23/2010 08:27 AM, Carlos Romero wrote:
> http://www.pjsip.org/
>    
PJSIP as a substitute for Sofia in high volume server applications? Are 
you a jester or a troll? :-)

All stacks have limitations, and Sofia has some. In the early days of FS 
development, every open source SIP stack was tried. All the alternatives 
were found to have far more limitations than Sofia.
> On Sat, May 22, 2010 at 3:53 PM, Sergey Okhapkin
> <sos at sokhapkin.dyndns.org>  wrote:
>    
>> Could you suggest a better open source SIP implementation?
>>
>> On Friday 21 May 2010, Carlos Romero wrote:
>>      
>>> why choose  a library that is hard to work with in the first place?
>>>
>>> it seems like sofia has many downsides... limited threads per
>>> sip-profile and now this
>>>
>>> On Fri, May 21, 2010 at 3:20 PM, Michael Collins<msc at freeswitch.org>
>>>        
>> wrote:
>>      
>>>> On Thu, May 20, 2010 at 11:46 PM, Carlos Romero<romerocarlos24 at
>>>>          
>> gmail.com>wrote:
>>      
>>>>> what do you mean that you don't have access to it?
>>>>>
>>>>> you have the code
>>>>>            
>>>> The SIP stack is supplied by Nokia. Open source or not, it is bad form
to
>>>> muck with someone else's library without due cause. Like all robust SIP
>>>> stacks, Sofia is huge, deep, complex, and otherwise uninviting of those
>>>> who would waltz in and do whatever they wish. If you feel that it is
even
>>>> within the realm of possibility then I recommend that you find the
>>>> location in the Sofia library where this takes place, make the
>>>> appropriate changes, and thoroughly test for regressions. Submit the
>>>> patch to the Sofia devs who will then evaluate it and see if it is
worthy
>>>> of inclusion.
>>>>
>>>> Or more simply put: "having the code" and "having access" are
COMPLETELY
>>>> different concepts.
>>>> -MC
>>>>
>>>>          
>>>>> On Thu, May 20, 2010 at 3:23 PM, Anthony Minessale
>>>>>
>>>>> <anthony.minessale at gmail.com>  wrote:
>>>>>            
>>>>>> not without intervention from the sofia devs,
>>>>>> Sofia sip does that deep down and we don't have access to it.
>>>>>> even when we supply it as a one big string it renders it out
depending
>>>>>>              
>>>>> on
>>>>>
>>>>>            
>>>>>> those settings.
>>>>>>              
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>            
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>          
>>> On Fri, May 21, 2010 at 4:34 AM, Carlos Romero<romerocarlos24 at gmail.com>
>>>        
>> wrote:
>>      
>>>> what do you mean that you don't have access to it?
>>>>
>>>> you have the code
>>>>
>>>> On Thu, May 20, 2010 at 3:23 PM, Anthony Minessale
>>>>
>>>> <anthony.minessale at gmail.com>  wrote:
>>>>          
>>>>> not without intervention from the sofia devs,
>>>>> Sofia sip does that deep down and we don't have access to it.
>>>>> even when we supply it as a one big string it renders it out depending
>>>>> on those settings.
>>>>>            
>>>
Steve


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



From freeswitch-list at puzzled.xs4all.nl  Mon May 24 03:50:58 2010
From: freeswitch-list at puzzled.xs4all.nl (Patrick)
Date: Mon, 24 May 2010 12:50:58 +0200
Subject: [Freeswitch-users] Mod_skinny: Call for testing!
In-Reply-To: <AANLkTik8hHRtOci8qTXmZVHOWvVxRg-OPm5zyRAMrWEe@mail.gmail.com>
References: <AANLkTik8hHRtOci8qTXmZVHOWvVxRg-OPm5zyRAMrWEe@mail.gmail.com>
Message-ID: <4BFA5A12.901@puzzled.xs4all.nl>

On 05/21/2010 02:54 AM, Mathieu Parent wrote:
[snip]

Hi Mathieu,

On Fedora 12 and FreeSWITCH git rev f89cbdd... (from yesterday) I get 
the following error:

quiet_libtool: compile:  gcc -DHAVE_CONFIG_H -I. 
-I../../../../src/include -I../../../../libs/xmlrpc-c 
-I/home/patrick/redhat/BUILD/freeswitch/src/include 
-I/home/patrick/redhat/BUILD/freeswitch/src/include 
-I/home/patrick/redhat/BUILD/freeswitch/libs/libteletone/src -fPIC 
-fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 
-I/home/patrick/redhat/BUILD/freeswitch/src/include 
-I/home/patrick/redhat/BUILD/freeswitch/src/include 
-I/home/patrick/redhat/BUILD/freeswitch/libs/libteletone/src -fPIC 
-Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 
-DHAVE_VISIBILITY=1 -g -ggdb -Wall -std=c99 -pedantic 
-Wdeclaration-after-statement -g -ggdb -MT 
mod_skinny_la-skinny_server.lo -MD -MP -MF 
.deps/mod_skinny_la-skinny_server.Tpo -c skinny_server.c  -fPIC -DPIC -o 
.libs/mod_skinny_la-skinny_server.o
cc1: warnings being treated as errors
skinny_server.c: In function ?skinny_handle_register?:
skinny_server.c:933: error: format ?%d? expects type ?int?, but argument 
11 has type ?long unsigned int?
skinny_server.c: In function ?skinny_handle_port_message?:
skinny_server.c:1084: error: format ?%d? expects type ?int?, but 
argument 11 has type ?long unsigned int?
skinny_server.c: In function ?skinny_handle_keypad_button_message?:
skinny_server.c:1104: error: format ?%d? expects type ?int?, but 
argument 11 has type ?long unsigned int?
skinny_server.c: In function ?skinny_handle_stimulus_message?:
skinny_server.c:1164: error: format ?%d? expects type ?int?, but 
argument 11 has type ?long unsigned int?
skinny_server.c: In function ?skinny_handle_speed_dial_stat_request?:
skinny_server.c:1271: error: format ?%d? expects type ?int?, but 
argument 11 has type ?long unsigned int?
skinny_server.c: In function ?skinny_handle_line_stat_request?:
skinny_server.c:1291: error: format ?%d? expects type ?int?, but 
argument 11 has type ?long unsigned int?
skinny_server.c: In function ?skinny_handle_capabilities_response?:
skinny_server.c:1501: error: format ?%d? expects type ?int?, but 
argument 11 has type ?long unsigned int?
skinny_server.c:1509: error: format ?%d? expects type ?int?, but 
argument 11 has type ?long unsigned int?
skinny_server.c: In function ?skinny_handle_alarm?:
skinny_server.c:1548: error: format ?%d? expects type ?int?, but 
argument 11 has type ?long unsigned int?
skinny_server.c: In function 
?skinny_handle_open_receive_channel_ack_message?:
skinny_server.c:1571: error: format ?%d? expects type ?int?, but 
argument 11 has type ?long unsigned int?
skinny_server.c: In function ?skinny_handle_soft_key_event_message?:
skinny_server.c:1722: error: format ?%d? expects type ?int?, but 
argument 11 has type ?long unsigned int?
skinny_server.c: In function ?skinny_headset_status_message?:
skinny_server.c:1847: error: format ?%d? expects type ?int?, but 
argument 11 has type ?long unsigned int?
skinny_server.c: In function 
?skinny_handle_register_available_lines_message?:
skinny_server.c:1855: error: format ?%d? expects type ?int?, but 
argument 11 has type ?long unsigned int?
skinny_server.c: In function ?skinny_handle_service_url_stat_request?:
skinny_server.c:1866: error: format ?%d? expects type ?int?, but 
argument 11 has type ?long unsigned int?
skinny_server.c: In function ?skinny_handle_feature_stat_request?:
skinny_server.c:1886: error: format ?%d? expects type ?int?, but 
argument 11 has type ?long unsigned int?
make: *** [mod_skinny_la-skinny_server.lo] Error 1

Regards,
Patrick


From tayeb.meftah at gmail.com  Tue May 25 03:03:10 2010
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Tue, 25 May 2010 12:03:10 +0200
Subject: [Freeswitch-users] did using easyroute
Message-ID: <4BFBA05E.8030906@gmail.com>

hello,
i see with easyroute i cabn route just one number and no a range
how do i route a range of numbers, like all number starting with 1747XXX 
to a special gateway, and 1748XXX to another?
thanks



From vetali100 at gmail.com  Mon May 24 05:12:38 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Mon, 24 May 2010 15:12:38 +0300
Subject: [Freeswitch-users] curl_xml
In-Reply-To: <1B2B2257-94F7-47BD-938A-369FB792D2D8@freeswitch.org>
References: <AANLkTiltmx-qzAAgkQjUcCAGsAMp9_X573zbQic24pf7@mail.gmail.com>
	<DF566DB4-AA3D-468C-A746-93F57BFED1D3@freeswitch.org>
	<AANLkTimFLCMqXYm27dq9oHPVtJ3pdldblhXcaOGu31BZ@mail.gmail.com>
	<03E6E212-1AF1-4CB3-8521-FBAD1DA53B7B@freeswitch.org>
	<AANLkTinDXvb_-69tK55KZIVurLjSkEP5cMrMTnXs7DXc@mail.gmail.com>
	<4BF9CAFC.7090302@solomo.de>
	<AANLkTin2KFDpjFdxDIYsocYg1bH0bIQsExKnrrOtHM73@mail.gmail.com>
	<1B2B2257-94F7-47BD-938A-369FB792D2D8@freeswitch.org>
Message-ID: <AANLkTinaUDJi5uXrollpVSXdgcpE7VNW79ikWcFdt07Y@mail.gmail.com>

Hi Erwin,
Maybe you need to add the first line <?xml version="1.0" encoding="UTF-8"
standalone="no"?>

This is what I have and it works perfectly, tested right now:

<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
  <section name="directory">
    <domain name="$${domain}">
      <params>
        <param name="dial-string" value="{presence_id=${dialed_user}@
${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
      </params>
      <groups>
        <group name="default">
          <users>
            <user id="1000">
              <params>
                <param name="a1-hash" value="somehashvalue"/>
                <param name="vm-a1-hash" value="somehashvalue"/>
              </params>
              <variables>
                <variable name="user_context" value="default"/>
                <variable name="accountcode" value="1000"/>
              </variables>
            </user>
          </users>
        </group>
      </groups>
    </domain>
  </section>
</document>

Regards,
Vitalie
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From jan.berger at video24.no  Mon May 24 05:45:32 2010
From: jan.berger at video24.no (Jan Berger)
Date: Mon, 24 May 2010 14:45:32 +0200
Subject: [Freeswitch-users] SCTP on Windows/C++/C#/VB
Message-ID: <9620701A18284477B3D2E5F32EE2A848@dell9400>

Hi,

 

I minor note in SCTP for those with special interests.

 

The SCTP on Windows is working, thought the current performance is only ca
30,000 messages per sec on my old laptop. The driver performs stable under
normal operations, but 100% CPU load over time will expose a snag that the
author currently are working to sort out.

 

I have tested on Windows XP Professional with an ordinary laptop. The stack
failed to install on Windows Vista 32 bit.

 

For all practical purposes the stack is sufficient for my development work,
but I will not recommend using in production yet. 

 

I have not tested the alternative which is www.sctp.de , but would like to
hear from anyone that has experience of this. 

 

If anyone is interested in this work let me know.

 

Jan

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From kris at kriskinc.com  Mon May 24 06:33:06 2010
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Mon, 24 May 2010 09:33:06 -0400
Subject: [Freeswitch-users] mod_dptools/redirect and 300 Multiple Choices
Message-ID: <87c5a4eafa458f9264629822e09e4683@mail.gmail.com>

Seriously... As the person who started this thread I'm disappointed to see
it come to this. Anyways, using mod_lcr and redirect I've been able to get
450 CPS (sustained) from FreeSWITCH on mediocre hardware. I'm more than
happy (read: thrilled) with that.

Shameless plug: I'll be telling the whole story of how I came to use FS as a
SIP redirect/LCR server on my blog in the next couple of days. I'll even
include the configs and SIPP scenario I used to test. Should be a good post!


--
Kristian Kielhofner
http://blog.krisk.org

----- Original Message -----
From: freeswitch-users-bounces at lists.freeswitch.org
<freeswitch-users-bounces at lists.freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
<freeswitch-users at lists.freeswitch.org>
Sent: Mon May 24 02:22:31 2010
Subject: Re: [Freeswitch-users] mod_dptools/redirect and 300 Multiple
Choices

wait until several hundred calls per second isn't enough for my needs and
then care about it?

This thread officially closed.  If anyone has any actual code to contribute
to this, they can do so on http://jira.freeswitch.org.

Mike

On May 24, 2010, at 2:03 AM, Carlos Romero wrote:

> Sorry I wasn't aware of this.
>
> So what do you intend to do to remove the limitations of sofia that
> seems to affect performance?
>
> Carlos Romero
>
> On Sat, May 22, 2010 at 6:59 PM, Steve Underwood wrote:
>> PJSIP as a substitute for Sofia in high volume server applications? Are
>> you a jester or a troll? :-)
>>
>> All stacks have limitations, and Sofia has some. In the early days of FS
>> development, every open source SIP stack was tried. All the alternatives
>> were found to have far more limitations than Sofia.
>


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


From peder at networkoblivion.com  Mon May 24 06:45:04 2010
From: peder at networkoblivion.com (Peder)
Date: Mon, 24 May 2010 08:45:04 -0500
Subject: [Freeswitch-users] xml_curl -  Binding has no url
Message-ID: <02cd01cafb47$5077aab0$f1670010$@com>

I am trying to use xml_curl with directory and it is failing with this
message:

2010-05-24 08:38:46.333352 [ERR] mod_xml_curl.c:444 Binding has no url!
2010-05-24 08:38:46.333352 [CRIT] switch_loadable_module.c:882 Error Loading
module /usr/local/freeswitch/mod/mod_xml_curl.so
**Module load routine returned an error**


I am sure I am missing something, but I can't figure out what it is.  The
module is built:

ls -al /usr/local/freeswitch/mod/mod_xml_curl.so

-rwxr-xr-x 1 root root 1498152 2010-05-24 08:23
/usr/local/freeswitch/mod/mod_xml_curl.so


The config file looks ok from what I can tell:

cat /usr/local/freeswitch/conf/autoload_configs/xml_curl.conf.xml

<configuration name="xml_curl.conf" description="cURL XML Gateway">
  <bindings>
    <binding name="directory_lookup">
    <param name="directory.php" value="http://localhost:80"
bindings="directory"/>
    </binding>
  </bindings>
</configuration>


I've stopped and started FS multiple times and still the same error on
startup and if I try and manually load the module with "load
mod_xml_curl.so".  Any ideas?  I am sure it is something dumb, but I just
can't see it.




From davis.erwin at gmail.com  Mon May 24 06:55:04 2010
From: davis.erwin at gmail.com (Erwin Davis)
Date: Mon, 24 May 2010 09:55:04 -0400
Subject: [Freeswitch-users] xml_curl - Binding has no url
In-Reply-To: <02cd01cafb47$5077aab0$f1670010$@com>
References: <02cd01cafb47$5077aab0$f1670010$@com>
Message-ID: <AANLkTimxg3jaXmWsE6qePkhrGjPcDj5vPXc8g6JZEWVM@mail.gmail.com>

I got the same error before. But once I ran the server on the configured
port and send back some NO result responses to the initial request from FS.
The error was gone.

On Mon, May 24, 2010 at 9:45 AM, Peder <peder at networkoblivion.com> wrote:

> I am trying to use xml_curl with directory and it is failing with this
> message:
>
> 2010-05-24 08:38:46.333352 [ERR] mod_xml_curl.c:444 Binding has no url!
> 2010-05-24 08:38:46.333352 [CRIT] switch_loadable_module.c:882 Error
> Loading
> module /usr/local/freeswitch/mod/mod_xml_curl.so
> **Module load routine returned an error**
>
>
> I am sure I am missing something, but I can't figure out what it is.  The
> module is built:
>
> ls -al /usr/local/freeswitch/mod/mod_xml_curl.so
>
> -rwxr-xr-x 1 root root 1498152 2010-05-24 08:23
> /usr/local/freeswitch/mod/mod_xml_curl.so
>
>
> The config file looks ok from what I can tell:
>
> cat /usr/local/freeswitch/conf/autoload_configs/xml_curl.conf.xml
>
> <configuration name="xml_curl.conf" description="cURL XML Gateway">
>  <bindings>
>    <binding name="directory_lookup">
>    <param name="directory.php" value="http://localhost:80"
> bindings="directory"/>
>    </binding>
>  </bindings>
> </configuration>
>
>
> I've stopped and started FS multiple times and still the same error on
> startup and if I try and manually load the module with "load
> mod_xml_curl.so".  Any ideas?  I am sure it is something dumb, but I just
> can't see it.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From davis.erwin at gmail.com  Mon May 24 06:57:51 2010
From: davis.erwin at gmail.com (Erwin Davis)
Date: Mon, 24 May 2010 09:57:51 -0400
Subject: [Freeswitch-users] curl_xml
In-Reply-To: <AANLkTinaUDJi5uXrollpVSXdgcpE7VNW79ikWcFdt07Y@mail.gmail.com>
References: <AANLkTiltmx-qzAAgkQjUcCAGsAMp9_X573zbQic24pf7@mail.gmail.com>
	<DF566DB4-AA3D-468C-A746-93F57BFED1D3@freeswitch.org>
	<AANLkTimFLCMqXYm27dq9oHPVtJ3pdldblhXcaOGu31BZ@mail.gmail.com>
	<03E6E212-1AF1-4CB3-8521-FBAD1DA53B7B@freeswitch.org>
	<AANLkTinDXvb_-69tK55KZIVurLjSkEP5cMrMTnXs7DXc@mail.gmail.com>
	<4BF9CAFC.7090302@solomo.de>
	<AANLkTin2KFDpjFdxDIYsocYg1bH0bIQsExKnrrOtHM73@mail.gmail.com>
	<1B2B2257-94F7-47BD-938A-369FB792D2D8@freeswitch.org>
	<AANLkTinaUDJi5uXrollpVSXdgcpE7VNW79ikWcFdt07Y@mail.gmail.com>
Message-ID: <AANLkTin61Y5yQWs--C1RggNo9ErnGqEYXhhSQ4XV68O1@mail.gmail.com>

Hi, I am stuck, My console log is in
http://pastebin.freeswitch.org/13030and my xml log is in
http://pastebin.freeswitch.org/13029.  I guess that the problem should be
pretty small but I just can not figure it out. -:(  Thanks for your help.

On Mon, May 24, 2010 at 8:12 AM, Vitalii Colosov <vetali100 at gmail.com>wrote:

> Hi Erwin,
> Maybe you need to add the first line <?xml version="1.0" encoding="UTF-8"
> standalone="no"?>
>
> This is what I have and it works perfectly, tested right now:
>
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
>   <section name="directory">
>     <domain name="$${domain}">
>       <params>
>         <param name="dial-string" value="{presence_id=${dialed_user}@
> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>       </params>
>       <groups>
>         <group name="default">
>           <users>
>             <user id="1000">
>               <params>
>                 <param name="a1-hash" value="somehashvalue"/>
>                 <param name="vm-a1-hash" value="somehashvalue"/>
>               </params>
>               <variables>
>                 <variable name="user_context" value="default"/>
>                 <variable name="accountcode" value="1000"/>
>               </variables>
>             </user>
>           </users>
>         </group>
>       </groups>
>     </domain>
>   </section>
> </document>
>
> Regards,
> Vitalie
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From peder at networkoblivion.com  Mon May 24 07:06:07 2010
From: peder at networkoblivion.com (Peder)
Date: Mon, 24 May 2010 09:06:07 -0500
Subject: [Freeswitch-users] xml_curl - Binding has no url
In-Reply-To: <AANLkTimxg3jaXmWsE6qePkhrGjPcDj5vPXc8g6JZEWVM@mail.gmail.com>
References: <02cd01cafb47$5077aab0$f1670010$@com>
	<AANLkTimxg3jaXmWsE6qePkhrGjPcDj5vPXc8g6JZEWVM@mail.gmail.com>
Message-ID: <02fb01cafb4a$41358a60$c3a09f20$@com>

The web server is running and the files are setup to return the correct
values, but I am not seeing any access to the webserver from FS.  If I bring
it up in my web browser, I see access_log entries, but nothing when FS
starts up.

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Erwin
Davis
Sent: Monday, May 24, 2010 8:55 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] xml_curl - Binding has no url

 

I got the same error before. But once I ran the server on the configured
port and send back some NO result responses to the initial request from FS.
The error was gone.

On Mon, May 24, 2010 at 9:45 AM, Peder <peder at networkoblivion.com> wrote:

I am trying to use xml_curl with directory and it is failing with this
message:

2010-05-24 08:38:46.333352 [ERR] mod_xml_curl.c:444 Binding has no url!
2010-05-24 08:38:46.333352 [CRIT] switch_loadable_module.c:882 Error Loading
module /usr/local/freeswitch/mod/mod_xml_curl.so
**Module load routine returned an error**


I am sure I am missing something, but I can't figure out what it is.  The
module is built:

ls -al /usr/local/freeswitch/mod/mod_xml_curl.so

-rwxr-xr-x 1 root root 1498152 2010-05-24 08:23
/usr/local/freeswitch/mod/mod_xml_curl.so


The config file looks ok from what I can tell:

cat /usr/local/freeswitch/conf/autoload_configs/xml_curl.conf.xml

<configuration name="xml_curl.conf" description="cURL XML Gateway">
 <bindings>
   <binding name="directory_lookup">
   <param name="directory.php" value="http://localhost:80"
bindings="directory"/>
   </binding>
 </bindings>
</configuration>


I've stopped and started FS multiple times and still the same error on
startup and if I try and manually load the module with "load
mod_xml_curl.so".  Any ideas?  I am sure it is something dumb, but I just
can't see it.



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

 

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From davis.erwin at gmail.com  Mon May 24 07:18:04 2010
From: davis.erwin at gmail.com (Erwin Davis)
Date: Mon, 24 May 2010 10:18:04 -0400
Subject: [Freeswitch-users] xml_curl - Binding has no url
In-Reply-To: <02fb01cafb4a$41358a60$c3a09f20$@com>
References: <02cd01cafb47$5077aab0$f1670010$@com>
	<AANLkTimxg3jaXmWsE6qePkhrGjPcDj5vPXc8g6JZEWVM@mail.gmail.com>
	<02fb01cafb4a$41358a60$c3a09f20$@com>
Message-ID: <AANLkTil3a7AgOgKkYUVQhElWnL_03B9EsjG4gyvssrEv@mail.gmail.com>

Hmmm, try to replace value = "http://localhost:80/directory.php".  My
understanding is that the value indicates the program to process the
incoming request.

On Mon, May 24, 2010 at 10:06 AM, Peder <peder at networkoblivion.com> wrote:

>  The web server is running and the files are setup to return the correct
> values, but I am not seeing any access to the webserver from FS.  If I bring
> it up in my web browser, I see access_log entries, but nothing when FS
> starts up.
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Erwin Davis
> *Sent:* Monday, May 24, 2010 8:55 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] xml_curl - Binding has no url
>
>
>
> I got the same error before. But once I ran the server on the configured
> port and send back some NO result responses to the initial request from FS.
> The error was gone.
>
> On Mon, May 24, 2010 at 9:45 AM, Peder <peder at networkoblivion.com> wrote:
>
> I am trying to use xml_curl with directory and it is failing with this
> message:
>
> 2010-05-24 08:38:46.333352 [ERR] mod_xml_curl.c:444 Binding has no url!
> 2010-05-24 08:38:46.333352 [CRIT] switch_loadable_module.c:882 Error
> Loading
> module /usr/local/freeswitch/mod/mod_xml_curl.so
> **Module load routine returned an error**
>
>
> I am sure I am missing something, but I can't figure out what it is.  The
> module is built:
>
> ls -al /usr/local/freeswitch/mod/mod_xml_curl.so
>
> -rwxr-xr-x 1 root root 1498152 2010-05-24 08:23
> /usr/local/freeswitch/mod/mod_xml_curl.so
>
>
> The config file looks ok from what I can tell:
>
> cat /usr/local/freeswitch/conf/autoload_configs/xml_curl.conf.xml
>
> <configuration name="xml_curl.conf" description="cURL XML Gateway">
>  <bindings>
>    <binding name="directory_lookup">
>    <param name="directory.php" value="http://localhost:80"
> bindings="directory"/>
>    </binding>
>  </bindings>
> </configuration>
>
>
> I've stopped and started FS multiple times and still the same error on
> startup and if I try and manually load the module with "load
> mod_xml_curl.so".  Any ideas?  I am sure it is something dumb, but I just
> can't see it.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From peder at networkoblivion.com  Mon May 24 07:34:15 2010
From: peder at networkoblivion.com (Peder)
Date: Mon, 24 May 2010 09:34:15 -0500
Subject: [Freeswitch-users] xml_curl - Binding has no url
In-Reply-To: <AANLkTil3a7AgOgKkYUVQhElWnL_03B9EsjG4gyvssrEv@mail.gmail.com>
References: <02cd01cafb47$5077aab0$f1670010$@com>	<AANLkTimxg3jaXmWsE6qePkhrGjPcDj5vPXc8g6JZEWVM@mail.gmail.com>	<02fb01cafb4a$41358a60$c3a09f20$@com>
	<AANLkTil3a7AgOgKkYUVQhElWnL_03B9EsjG4gyvssrEv@mail.gmail.com>
Message-ID: <032c01cafb4e$2fa6ec90$8ef4c5b0$@com>

Same result.  I can telnet from the box itself and I get a result, although
I am not seeing the fields filled up as I am not sending any info in the
request, but it confirms that the web server is responding.  Again, note
that when FS loads, or I try to manually load the module, there are no
entries in the web server logs indicating that FS is even requesting info
from the web server.

 

 

root at AS-TEST-A:/usr/local/freeswitch/conf/autoload_configs# telnet localhost
80

 

Trying 127.0.0.1...

Connected to localhost.

Escape character is '^]'.

get /directory.php http/1.0

 

HTTP/1.1 200 OK

Date: Mon, 24 May 2010 14:31:49 GMT

Server: Apache/2.2.8 (Ubuntu) PHP/5.2.4-2ubuntu5.9 with Suhosin-Patch

X-Powered-By: PHP/5.2.4-2ubuntu5.9

Content-Length: 898

Connection: close

Content-Type: text/html

 

Content-Type: text/xml

<?xml version="1.0"?>

<document type="freeswitch/xml">

<section name="directory" description="example_curl_directory">

  <domain name="192.168.1.108">

    <user id="" mailbox="">

      <params>

        <param name="dial-string"
value="${sofia_contact(${dialed_user}@${dialed_domain})}"/>

        <param name="password" value=""/>

        <param name="vm-password" value=""/>

        <param name="email-addr" value=""/>

        <param name="vm-mailto" value=""/>

        <param name="vm-delete-file" value="1"/>

        <param name="vm-attach-file" value="1"/>

        <param name="vm-keep-local-after-email" value="0"/>

        <param name="vm-email-all-messages" value="1"/>

      </params>

      <variables>

        <variable name="user_context" value=""/>

        <variable name="accountcode" value=""/>

      </variables>

    </user>

  </domain>

</section>

</document>

Connection closed by foreign host.

 

 

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Erwin
Davis
Sent: Monday, May 24, 2010 9:18 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] xml_curl - Binding has no url

 

Hmmm, try to replace value = "http://localhost:80/directory.php".  My
understanding is that the value indicates the program to process the
incoming request. 

On Mon, May 24, 2010 at 10:06 AM, Peder <peder at networkoblivion.com> wrote:

The web server is running and the files are setup to return the correct
values, but I am not seeing any access to the webserver from FS.  If I bring
it up in my web browser, I see access_log entries, but nothing when FS
starts up.

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Erwin
Davis
Sent: Monday, May 24, 2010 8:55 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] xml_curl - Binding has no url

 

I got the same error before. But once I ran the server on the configured
port and send back some NO result responses to the initial request from FS.
The error was gone.

On Mon, May 24, 2010 at 9:45 AM, Peder <peder at networkoblivion.com> wrote:

I am trying to use xml_curl with directory and it is failing with this
message:

2010-05-24 08:38:46.333352 [ERR] mod_xml_curl.c:444 Binding has no url!
2010-05-24 08:38:46.333352 [CRIT] switch_loadable_module.c:882 Error Loading
module /usr/local/freeswitch/mod/mod_xml_curl.so
**Module load routine returned an error**


I am sure I am missing something, but I can't figure out what it is.  The
module is built:

ls -al /usr/local/freeswitch/mod/mod_xml_curl.so

-rwxr-xr-x 1 root root 1498152 2010-05-24 08:23
/usr/local/freeswitch/mod/mod_xml_curl.so


The config file looks ok from what I can tell:

cat /usr/local/freeswitch/conf/autoload_configs/xml_curl.conf.xml

<configuration name="xml_curl.conf" description="cURL XML Gateway">
 <bindings>
   <binding name="directory_lookup">
   <param name="directory.php" value="http://localhost:80"
bindings="directory"/>
   </binding>
 </bindings>
</configuration>


I've stopped and started FS multiple times and still the same error on
startup and if I try and manually load the module with "load
mod_xml_curl.so".  Any ideas?  I am sure it is something dumb, but I just
can't see it.



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

 


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

 

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From testeador01 at gmail.com  Mon May 24 07:34:49 2010
From: testeador01 at gmail.com (Milena)
Date: Mon, 24 May 2010 09:34:49 -0500
Subject: [Freeswitch-users] curl_xml
In-Reply-To: <AANLkTin61Y5yQWs--C1RggNo9ErnGqEYXhhSQ4XV68O1@mail.gmail.com>
References: <AANLkTiltmx-qzAAgkQjUcCAGsAMp9_X573zbQic24pf7@mail.gmail.com>
	<DF566DB4-AA3D-468C-A746-93F57BFED1D3@freeswitch.org>
	<AANLkTimFLCMqXYm27dq9oHPVtJ3pdldblhXcaOGu31BZ@mail.gmail.com>
	<03E6E212-1AF1-4CB3-8521-FBAD1DA53B7B@freeswitch.org>
	<AANLkTinDXvb_-69tK55KZIVurLjSkEP5cMrMTnXs7DXc@mail.gmail.com>
	<4BF9CAFC.7090302@solomo.de>
	<AANLkTin2KFDpjFdxDIYsocYg1bH0bIQsExKnrrOtHM73@mail.gmail.com>
	<1B2B2257-94F7-47BD-938A-369FB792D2D8@freeswitch.org>
	<AANLkTinaUDJi5uXrollpVSXdgcpE7VNW79ikWcFdt07Y@mail.gmail.com>
	<AANLkTin61Y5yQWs--C1RggNo9ErnGqEYXhhSQ4XV68O1@mail.gmail.com>
Message-ID: <AANLkTim6sWASDmj8w6KF9gsgB8joPKZC6VX34HpEOCdE@mail.gmail.com>

Hello,
You're not meant to send a whole website full of html tags as a response.
Just reply the part of the lines 29 - 46 on your pastebin and get rid of the
rest.


2010,
/5/24 Erwin Davis <davis.erwin at gmail.com>

> Hi, I am stuck, My console log is in http://pastebin.freeswitch.org/13030and my xml log is in
> http://pastebin.freeswitch.org/13029.  I guess that the problem should be
> pretty small but I just can not figure it out. -:(  Thanks for your help.
>
> On Mon, May 24, 2010 at 8:12 AM, Vitalii Colosov <vetali100 at gmail.com>wrote:
>
>> Hi Erwin,
>> Maybe you need to add the first line <?xml version="1.0" encoding="UTF-8"
>> standalone="no"?>
>>
>> This is what I have and it works perfectly, tested right now:
>>
>> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>> <document type="freeswitch/xml">
>>   <section name="directory">
>>     <domain name="$${domain}">
>>       <params>
>>         <param name="dial-string" value="{presence_id=${dialed_user}@
>> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>>       </params>
>>       <groups>
>>         <group name="default">
>>           <users>
>>             <user id="1000">
>>               <params>
>>                 <param name="a1-hash" value="somehashvalue"/>
>>                 <param name="vm-a1-hash" value="somehashvalue"/>
>>               </params>
>>               <variables>
>>                 <variable name="user_context" value="default"/>
>>                 <variable name="accountcode" value="1000"/>
>>               </variables>
>>             </user>
>>           </users>
>>         </group>
>>       </groups>
>>     </domain>
>>   </section>
>> </document>
>>
>> Regards,
>> Vitalie
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From peder at networkoblivion.com  Mon May 24 07:46:26 2010
From: peder at networkoblivion.com (Peder)
Date: Mon, 24 May 2010 09:46:26 -0500
Subject: [Freeswitch-users] xml_curl - Binding has no url
In-Reply-To: <02fb01cafb4a$41358a60$c3a09f20$@com>
References: <02cd01cafb47$5077aab0$f1670010$@com>	<AANLkTimxg3jaXmWsE6qePkhrGjPcDj5vPXc8g6JZEWVM@mail.gmail.com>
	<02fb01cafb4a$41358a60$c3a09f20$@com>
Message-ID: <034601cafb4f$e2f16720$a8d43560$@com>

I figured it out.  I was changing "gateway-url" to the actual url name.  It
is actually a keyword.  I knew it was something dumb..   Here is the correct
syntax for what I had:

 

<param name="gateway-url" value="http://localhost/directory.php"
bindings="directory"/>

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peder
Sent: Monday, May 24, 2010 9:06 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] xml_curl - Binding has no url

 

The web server is running and the files are setup to return the correct
values, but I am not seeing any access to the webserver from FS.  If I bring
it up in my web browser, I see access_log entries, but nothing when FS
starts up.

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Erwin
Davis
Sent: Monday, May 24, 2010 8:55 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] xml_curl - Binding has no url

 

I got the same error before. But once I ran the server on the configured
port and send back some NO result responses to the initial request from FS.
The error was gone.

On Mon, May 24, 2010 at 9:45 AM, Peder <peder at networkoblivion.com> wrote:

I am trying to use xml_curl with directory and it is failing with this
message:

2010-05-24 08:38:46.333352 [ERR] mod_xml_curl.c:444 Binding has no url!
2010-05-24 08:38:46.333352 [CRIT] switch_loadable_module.c:882 Error Loading
module /usr/local/freeswitch/mod/mod_xml_curl.so
**Module load routine returned an error**


I am sure I am missing something, but I can't figure out what it is.  The
module is built:

ls -al /usr/local/freeswitch/mod/mod_xml_curl.so

-rwxr-xr-x 1 root root 1498152 2010-05-24 08:23
/usr/local/freeswitch/mod/mod_xml_curl.so


The config file looks ok from what I can tell:

cat /usr/local/freeswitch/conf/autoload_configs/xml_curl.conf.xml

<configuration name="xml_curl.conf" description="cURL XML Gateway">
 <bindings>
   <binding name="directory_lookup">
   <param name="directory.php" value="http://localhost:80"
bindings="directory"/>
   </binding>
 </bindings>
</configuration>


I've stopped and started FS multiple times and still the same error on
startup and if I try and manually load the module with "load
mod_xml_curl.so".  Any ideas?  I am sure it is something dumb, but I just
can't see it.



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

 

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From testeador01 at gmail.com  Mon May 24 07:46:10 2010
From: testeador01 at gmail.com (Milena)
Date: Mon, 24 May 2010 09:46:10 -0500
Subject: [Freeswitch-users] curl_xml
In-Reply-To: <AANLkTim6sWASDmj8w6KF9gsgB8joPKZC6VX34HpEOCdE@mail.gmail.com>
References: <AANLkTiltmx-qzAAgkQjUcCAGsAMp9_X573zbQic24pf7@mail.gmail.com>
	<AANLkTimFLCMqXYm27dq9oHPVtJ3pdldblhXcaOGu31BZ@mail.gmail.com>
	<03E6E212-1AF1-4CB3-8521-FBAD1DA53B7B@freeswitch.org>
	<AANLkTinDXvb_-69tK55KZIVurLjSkEP5cMrMTnXs7DXc@mail.gmail.com>
	<4BF9CAFC.7090302@solomo.de>
	<AANLkTin2KFDpjFdxDIYsocYg1bH0bIQsExKnrrOtHM73@mail.gmail.com>
	<1B2B2257-94F7-47BD-938A-369FB792D2D8@freeswitch.org>
	<AANLkTinaUDJi5uXrollpVSXdgcpE7VNW79ikWcFdt07Y@mail.gmail.com>
	<AANLkTin61Y5yQWs--C1RggNo9ErnGqEYXhhSQ4XV68O1@mail.gmail.com>
	<AANLkTim6sWASDmj8w6KF9gsgB8joPKZC6VX34HpEOCdE@mail.gmail.com>
Message-ID: <AANLkTimQwj2rYiuXSOsQDuHR1qPI21aFEVWCFwYbiK_6@mail.gmail.com>

To debug of what's being returned, type this on the CLI:
find_user_xml id 1018 192.168.1.31


2010/5/24 Milena <testeador01 at gmail.com>

> Hello,
> You're not meant to send a whole website full of html tags as a response.
> Just reply the part of the lines 29 - 46 on your pastebin and get rid of
> the rest.
>
>
> 2010,
> /5/24 Erwin Davis <davis.erwin at gmail.com>
>
> Hi, I am stuck, My console log is in http://pastebin.freeswitch.org/13030and my xml log is in
>> http://pastebin.freeswitch.org/13029.  I guess that the problem should be
>> pretty small but I just can not figure it out. -:(  Thanks for your help.
>>
>> On Mon, May 24, 2010 at 8:12 AM, Vitalii Colosov <vetali100 at gmail.com>wrote:
>>
>>> Hi Erwin,
>>> Maybe you need to add the first line <?xml version="1.0" encoding="UTF-8"
>>> standalone="no"?>
>>>
>>> This is what I have and it works perfectly, tested right now:
>>>
>>> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>>> <document type="freeswitch/xml">
>>>   <section name="directory">
>>>     <domain name="$${domain}">
>>>       <params>
>>>         <param name="dial-string" value="{presence_id=${dialed_user}@
>>> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>>>       </params>
>>>       <groups>
>>>         <group name="default">
>>>           <users>
>>>             <user id="1000">
>>>               <params>
>>>                 <param name="a1-hash" value="somehashvalue"/>
>>>                 <param name="vm-a1-hash" value="somehashvalue"/>
>>>               </params>
>>>               <variables>
>>>                 <variable name="user_context" value="default"/>
>>>                 <variable name="accountcode" value="1000"/>
>>>               </variables>
>>>             </user>
>>>           </users>
>>>         </group>
>>>       </groups>
>>>     </domain>
>>>   </section>
>>> </document>
>>>
>>> Regards,
>>> Vitalie
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From davis.erwin at gmail.com  Mon May 24 07:52:26 2010
From: davis.erwin at gmail.com (Erwin Davis)
Date: Mon, 24 May 2010 10:52:26 -0400
Subject: [Freeswitch-users] curl_xml
In-Reply-To: <AANLkTim6sWASDmj8w6KF9gsgB8joPKZC6VX34HpEOCdE@mail.gmail.com>
References: <AANLkTiltmx-qzAAgkQjUcCAGsAMp9_X573zbQic24pf7@mail.gmail.com>
	<AANLkTimFLCMqXYm27dq9oHPVtJ3pdldblhXcaOGu31BZ@mail.gmail.com>
	<03E6E212-1AF1-4CB3-8521-FBAD1DA53B7B@freeswitch.org>
	<AANLkTinDXvb_-69tK55KZIVurLjSkEP5cMrMTnXs7DXc@mail.gmail.com>
	<4BF9CAFC.7090302@solomo.de>
	<AANLkTin2KFDpjFdxDIYsocYg1bH0bIQsExKnrrOtHM73@mail.gmail.com>
	<1B2B2257-94F7-47BD-938A-369FB792D2D8@freeswitch.org>
	<AANLkTinaUDJi5uXrollpVSXdgcpE7VNW79ikWcFdt07Y@mail.gmail.com>
	<AANLkTin61Y5yQWs--C1RggNo9ErnGqEYXhhSQ4XV68O1@mail.gmail.com>
	<AANLkTim6sWASDmj8w6KF9gsgB8joPKZC6VX34HpEOCdE@mail.gmail.com>
Message-ID: <AANLkTimWDqSf2a7ZacI6q_4AQP5wm1sJImEKZBX3QzbC@mail.gmail.com>

yeah, my code replied with line 29-46. I donot know where the rest of html
code was generated.

On Mon, May 24, 2010 at 10:34 AM, Milena <testeador01 at gmail.com> wrote:

> Hello,
> You're not meant to send a whole website full of html tags as a response.
> Just reply the part of the lines 29 - 46 on your pastebin and get rid of
> the rest.
>
>
> 2010,
> /5/24 Erwin Davis <davis.erwin at gmail.com>
>
> Hi, I am stuck, My console log is in http://pastebin.freeswitch.org/13030and my xml log is in
>> http://pastebin.freeswitch.org/13029.  I guess that the problem should be
>> pretty small but I just can not figure it out. -:(  Thanks for your help.
>>
>> On Mon, May 24, 2010 at 8:12 AM, Vitalii Colosov <vetali100 at gmail.com>wrote:
>>
>>> Hi Erwin,
>>> Maybe you need to add the first line <?xml version="1.0" encoding="UTF-8"
>>> standalone="no"?>
>>>
>>> This is what I have and it works perfectly, tested right now:
>>>
>>> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>>> <document type="freeswitch/xml">
>>>   <section name="directory">
>>>     <domain name="$${domain}">
>>>       <params>
>>>         <param name="dial-string" value="{presence_id=${dialed_user}@
>>> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>>>       </params>
>>>       <groups>
>>>         <group name="default">
>>>           <users>
>>>             <user id="1000">
>>>               <params>
>>>                 <param name="a1-hash" value="somehashvalue"/>
>>>                 <param name="vm-a1-hash" value="somehashvalue"/>
>>>               </params>
>>>               <variables>
>>>                 <variable name="user_context" value="default"/>
>>>                 <variable name="accountcode" value="1000"/>
>>>               </variables>
>>>             </user>
>>>           </users>
>>>         </group>
>>>       </groups>
>>>     </domain>
>>>   </section>
>>> </document>
>>>
>>> Regards,
>>> Vitalie
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From vetali100 at gmail.com  Mon May 24 08:00:57 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Mon, 24 May 2010 18:00:57 +0300
Subject: [Freeswitch-users] curl_xml
In-Reply-To: <AANLkTimWDqSf2a7ZacI6q_4AQP5wm1sJImEKZBX3QzbC@mail.gmail.com>
References: <AANLkTiltmx-qzAAgkQjUcCAGsAMp9_X573zbQic24pf7@mail.gmail.com>
	<03E6E212-1AF1-4CB3-8521-FBAD1DA53B7B@freeswitch.org>
	<AANLkTinDXvb_-69tK55KZIVurLjSkEP5cMrMTnXs7DXc@mail.gmail.com>
	<4BF9CAFC.7090302@solomo.de>
	<AANLkTin2KFDpjFdxDIYsocYg1bH0bIQsExKnrrOtHM73@mail.gmail.com>
	<1B2B2257-94F7-47BD-938A-369FB792D2D8@freeswitch.org>
	<AANLkTinaUDJi5uXrollpVSXdgcpE7VNW79ikWcFdt07Y@mail.gmail.com>
	<AANLkTin61Y5yQWs--C1RggNo9ErnGqEYXhhSQ4XV68O1@mail.gmail.com>
	<AANLkTim6sWASDmj8w6KF9gsgB8joPKZC6VX34HpEOCdE@mail.gmail.com>
	<AANLkTimWDqSf2a7ZacI6q_4AQP5wm1sJImEKZBX3QzbC@mail.gmail.com>
Message-ID: <AANLkTilAIOnMu1bmJBOTsCvrnSfytRE8tenp6JEVTMc-@mail.gmail.com>

Please try to use my attached php script, if you can.

I changed it to fit your data (user 1018, pass: 1018), but I did not check
for any syntax error (hope it does not have :-) ).

Please reply back on result.

Regards,

Vitalie

2010/5/24 Erwin Davis <davis.erwin at gmail.com>

> yeah, my code replied with line 29-46. I donot know where the rest of html
> code was generated.
>
> On Mon, May 24, 2010 at 10:34 AM, Milena <testeador01 at gmail.com> wrote:
>
>> Hello,
>> You're not meant to send a whole website full of html tags as a response.
>> Just reply the part of the lines 29 - 46 on your pastebin and get rid of
>> the rest.
>>
>>
>> 2010,
>> /5/24 Erwin Davis <davis.erwin at gmail.com>
>>
>> Hi, I am stuck, My console log is in http://pastebin.freeswitch.org/13030and my xml log is in
>>> http://pastebin.freeswitch.org/13029.  I guess that the problem should
>>> be pretty small but I just can not figure it out. -:(  Thanks for your help.
>>>
>>> On Mon, May 24, 2010 at 8:12 AM, Vitalii Colosov <vetali100 at gmail.com>wrote:
>>>
>>>> Hi Erwin,
>>>> Maybe you need to add the first line <?xml version="1.0"
>>>> encoding="UTF-8" standalone="no"?>
>>>>
>>>> This is what I have and it works perfectly, tested right now:
>>>>
>>>> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>>>> <document type="freeswitch/xml">
>>>>   <section name="directory">
>>>>     <domain name="$${domain}">
>>>>       <params>
>>>>         <param name="dial-string" value="{presence_id=${dialed_user}@
>>>> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>>>>       </params>
>>>>       <groups>
>>>>         <group name="default">
>>>>           <users>
>>>>             <user id="1000">
>>>>               <params>
>>>>                 <param name="a1-hash" value="somehashvalue"/>
>>>>                 <param name="vm-a1-hash" value="somehashvalue"/>
>>>>               </params>
>>>>               <variables>
>>>>                 <variable name="user_context" value="default"/>
>>>>                 <variable name="accountcode" value="1000"/>
>>>>               </variables>
>>>>             </user>
>>>>           </users>
>>>>         </group>
>>>>       </groups>
>>>>     </domain>
>>>>   </section>
>>>> </document>
>>>>
>>>> Regards,
>>>> Vitalie
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From vetali100 at gmail.com  Mon May 24 08:14:33 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Mon, 24 May 2010 18:14:33 +0300
Subject: [Freeswitch-users] curl_xml
In-Reply-To: <AANLkTilAIOnMu1bmJBOTsCvrnSfytRE8tenp6JEVTMc-@mail.gmail.com>
References: <AANLkTiltmx-qzAAgkQjUcCAGsAMp9_X573zbQic24pf7@mail.gmail.com>
	<AANLkTinDXvb_-69tK55KZIVurLjSkEP5cMrMTnXs7DXc@mail.gmail.com>
	<4BF9CAFC.7090302@solomo.de>
	<AANLkTin2KFDpjFdxDIYsocYg1bH0bIQsExKnrrOtHM73@mail.gmail.com>
	<1B2B2257-94F7-47BD-938A-369FB792D2D8@freeswitch.org>
	<AANLkTinaUDJi5uXrollpVSXdgcpE7VNW79ikWcFdt07Y@mail.gmail.com>
	<AANLkTin61Y5yQWs--C1RggNo9ErnGqEYXhhSQ4XV68O1@mail.gmail.com>
	<AANLkTim6sWASDmj8w6KF9gsgB8joPKZC6VX34HpEOCdE@mail.gmail.com>
	<AANLkTimWDqSf2a7ZacI6q_4AQP5wm1sJImEKZBX3QzbC@mail.gmail.com>
	<AANLkTilAIOnMu1bmJBOTsCvrnSfytRE8tenp6JEVTMc-@mail.gmail.com>
Message-ID: <AANLkTin9OKJNURCKZ7__EMGyo1_4WYXufdQhDmYAcJOb@mail.gmail.com>

Sorry there was an else related to the missing entry, i removed it.

Please try another one.

2010/5/24 Vitalii Colosov <vetali100 at gmail.com>

> Please try to use my attached php script, if you can.
>
> I changed it to fit your data (user 1018, pass: 1018), but I did not check
> for any syntax error (hope it does not have :-) ).
>
> Please reply back on result.
>
> Regards,
>
> Vitalie
>
>
> 2010/5/24 Erwin Davis <davis.erwin at gmail.com>
>
>> yeah, my code replied with line 29-46. I donot know where the rest of html
>> code was generated.
>>
>> On Mon, May 24, 2010 at 10:34 AM, Milena <testeador01 at gmail.com> wrote:
>>
>>> Hello,
>>> You're not meant to send a whole website full of html tags as a response.
>>> Just reply the part of the lines 29 - 46 on your pastebin and get rid of
>>> the rest.
>>>
>>>
>>> 2010,
>>> /5/24 Erwin Davis <davis.erwin at gmail.com>
>>>
>>> Hi, I am stuck, My console log is in
>>>> http://pastebin.freeswitch.org/13030 and my xml log is in
>>>> http://pastebin.freeswitch.org/13029.  I guess that the problem should
>>>> be pretty small but I just can not figure it out. -:(  Thanks for your help.
>>>>
>>>> On Mon, May 24, 2010 at 8:12 AM, Vitalii Colosov <vetali100 at gmail.com>wrote:
>>>>
>>>>> Hi Erwin,
>>>>> Maybe you need to add the first line <?xml version="1.0"
>>>>> encoding="UTF-8" standalone="no"?>
>>>>>
>>>>> This is what I have and it works perfectly, tested right now:
>>>>>
>>>>> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>>>>> <document type="freeswitch/xml">
>>>>>   <section name="directory">
>>>>>     <domain name="$${domain}">
>>>>>       <params>
>>>>>         <param name="dial-string" value="{presence_id=${dialed_user}@
>>>>> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>>>>>       </params>
>>>>>       <groups>
>>>>>         <group name="default">
>>>>>           <users>
>>>>>             <user id="1000">
>>>>>               <params>
>>>>>                 <param name="a1-hash" value="somehashvalue"/>
>>>>>                 <param name="vm-a1-hash" value="somehashvalue"/>
>>>>>               </params>
>>>>>               <variables>
>>>>>                 <variable name="user_context" value="default"/>
>>>>>                 <variable name="accountcode" value="1000"/>
>>>>>               </variables>
>>>>>             </user>
>>>>>           </users>
>>>>>         </group>
>>>>>       </groups>
>>>>>     </domain>
>>>>>   </section>
>>>>> </document>
>>>>>
>>>>> Regards,
>>>>> Vitalie
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From vetali100 at gmail.com  Mon May 24 08:38:49 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Mon, 24 May 2010 18:38:49 +0300
Subject: [Freeswitch-users] curl_xml
In-Reply-To: <AANLkTin9OKJNURCKZ7__EMGyo1_4WYXufdQhDmYAcJOb@mail.gmail.com>
References: <AANLkTiltmx-qzAAgkQjUcCAGsAMp9_X573zbQic24pf7@mail.gmail.com>
	<4BF9CAFC.7090302@solomo.de>
	<AANLkTin2KFDpjFdxDIYsocYg1bH0bIQsExKnrrOtHM73@mail.gmail.com>
	<1B2B2257-94F7-47BD-938A-369FB792D2D8@freeswitch.org>
	<AANLkTinaUDJi5uXrollpVSXdgcpE7VNW79ikWcFdt07Y@mail.gmail.com>
	<AANLkTin61Y5yQWs--C1RggNo9ErnGqEYXhhSQ4XV68O1@mail.gmail.com>
	<AANLkTim6sWASDmj8w6KF9gsgB8joPKZC6VX34HpEOCdE@mail.gmail.com>
	<AANLkTimWDqSf2a7ZacI6q_4AQP5wm1sJImEKZBX3QzbC@mail.gmail.com>
	<AANLkTilAIOnMu1bmJBOTsCvrnSfytRE8tenp6JEVTMc-@mail.gmail.com>
	<AANLkTin9OKJNURCKZ7__EMGyo1_4WYXufdQhDmYAcJOb@mail.gmail.com>
Message-ID: <AANLkTimR6zWJX3bwGCuvt5efDRlj3qY_Xiisn3doGupD@mail.gmail.com>

Just to avoid any misunderstanding - this is to check all other things
around freeswitch configuration.

If this will work, it will be very easy to add database lookup for the user
trying to register:


$user = $_POST['user'];
$query="select password from your_users_table where user ='$user'";
$result=mysqli_query($db_connection, $query);

...get user and password from the result

And then use $user and $password variables instead of hard-coded "1018",
"1018" in the code I provided.

Really hope this helps, it works for me.

Regards,
Vitalie


2010/5/24 Vitalii Colosov <vetali100 at gmail.com>

> Sorry there was an else related to the missing entry, i removed it.
>
> Please try another one.
>
> 2010/5/24 Vitalii Colosov <vetali100 at gmail.com>
>
> Please try to use my attached php script, if you can.
>>
>> I changed it to fit your data (user 1018, pass: 1018), but I did not check
>> for any syntax error (hope it does not have :-) ).
>>
>> Please reply back on result.
>>
>> Regards,
>>
>> Vitalie
>>
>>
>> 2010/5/24 Erwin Davis <davis.erwin at gmail.com>
>>
>>> yeah, my code replied with line 29-46. I donot know where the rest of
>>> html code was generated.
>>>
>>> On Mon, May 24, 2010 at 10:34 AM, Milena <testeador01 at gmail.com> wrote:
>>>
>>>> Hello,
>>>> You're not meant to send a whole website full of html tags as a
>>>> response.
>>>> Just reply the part of the lines 29 - 46 on your pastebin and get rid of
>>>> the rest.
>>>>
>>>>
>>>> 2010,
>>>> /5/24 Erwin Davis <davis.erwin at gmail.com>
>>>>
>>>> Hi, I am stuck, My console log is in
>>>>> http://pastebin.freeswitch.org/13030 and my xml log is in
>>>>> http://pastebin.freeswitch.org/13029.  I guess that the problem should
>>>>> be pretty small but I just can not figure it out. -:(  Thanks for your help.
>>>>>
>>>>> On Mon, May 24, 2010 at 8:12 AM, Vitalii Colosov <vetali100 at gmail.com>wrote:
>>>>>
>>>>>> Hi Erwin,
>>>>>> Maybe you need to add the first line <?xml version="1.0"
>>>>>> encoding="UTF-8" standalone="no"?>
>>>>>>
>>>>>> This is what I have and it works perfectly, tested right now:
>>>>>>
>>>>>> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>>>>>> <document type="freeswitch/xml">
>>>>>>   <section name="directory">
>>>>>>     <domain name="$${domain}">
>>>>>>       <params>
>>>>>>         <param name="dial-string" value="{presence_id=${dialed_user}@
>>>>>> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>>>>>>       </params>
>>>>>>       <groups>
>>>>>>         <group name="default">
>>>>>>           <users>
>>>>>>             <user id="1000">
>>>>>>               <params>
>>>>>>                 <param name="a1-hash" value="somehashvalue"/>
>>>>>>                 <param name="vm-a1-hash" value="somehashvalue"/>
>>>>>>               </params>
>>>>>>               <variables>
>>>>>>                 <variable name="user_context" value="default"/>
>>>>>>                 <variable name="accountcode" value="1000"/>
>>>>>>               </variables>
>>>>>>             </user>
>>>>>>           </users>
>>>>>>         </group>
>>>>>>       </groups>
>>>>>>     </domain>
>>>>>>   </section>
>>>>>> </document>
>>>>>>
>>>>>> Regards,
>>>>>> Vitalie
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
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From wahalla at gmail.com  Sun May 23 01:51:41 2010
From: wahalla at gmail.com (Wa Halla)
Date: Sun, 23 May 2010 09:51:41 +0100
Subject: [Freeswitch-users] make a remote extension call from a particular
	gateway
Message-ID: <AANLkTikAB5oG3ujobJ8iv6L61JSqErAM8QK7YpUhNm64@mail.gmail.com>

I have a branch office that I want to call from a particular gateway. The
branch is on extension 1008. With the dialplan below any extension uses
gateway4 even extension 1008

<extension name="domestic.uk">
    <condition field="${toll_allow}" expression="domestic"/>
    <condition field="destination_number" expression="^(\d{11})$">
      <action application="set"
data="effective_caller_id_number=${outbound_caller_id_number}"/>
      <action application="set"
data="effective_caller_id_name=${outbound_caller_id_name}"/>
      <action application="bridge" data="sofia/gateway/gateway4/$1"/>
    </condition>
  </extension>

<extension name="domestic1008.uk">
    <condition field="${toll_allow}" expression="domestic"/>
    <condition field="sip_auth_username" expression="^(1008)$"/>
    <condition field="destination_number" expression="^(\d{11})$">
      <action application="set"
data="effective_caller_id_number=${outbound_caller_id_number}"/>
      <action application="set"
data="effective_caller_id_name=${outbound_caller_id_name}"/>
      <action application="bridge" data="sofia/gateway/gateway1/$1"/>
    </condition>
  </extension>

Any suggestions of what I am doing wrong. I have tried <condition
field="sip_auth_username" expression="^(1008)$"/> with and <condition
field="sip_auth_username" expression="^1008$"/> without the brackets so have
elimated that.

If I switch the gateways in the  domestic.uk extension, all calls go from
gateway1. So it seems the domestic1008.uk never gets executed as true.
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From dswardstrom at remotelink.com  Mon May 24 09:07:42 2010
From: dswardstrom at remotelink.com (David Swardstrom)
Date: Mon, 24 May 2010 11:07:42 -0500 (CDT)
Subject: [Freeswitch-users] Searching mailing lists
Message-ID: <1229079085.100.1274717262156.JavaMail.root@srvr12.remotelinkml.com>

I am new to using FreeSwitch but I have a couple of questions about things.
But before I send in the questions, I would like to see if there is any 
discussion either in the Development or the User list.
However, there does not seem to be a search engine capability supplied.
Is there a search capability?
If so, is there a web page for this?

Regards,
Paul David Swardstrom



From dswardstrom at remotelink.com  Mon May 24 11:22:01 2010
From: dswardstrom at remotelink.com (David Swardstrom)
Date: Mon, 24 May 2010 13:22:01 -0500 (CDT)
Subject: [Freeswitch-users] Performing system wide initialization during
	startup
Message-ID: <833424957.122.1274725321624.JavaMail.root@srvr12.remotelinkml.com>

I am recreating a conferencing application on FreeSwitch to replace a proprietary application 
originally developed using Dialogic ISDN hardware.
I am using mod_conference and a JavaScript application based on the JavaScript samples.

As part of this application, I have to track some counters and other data that are global
to all of the conference/channel instances. I would like to initialize this code during
startup. I could do this multiple ways including using C or C++ code. 
I would prefer to accomplish this by running a small JavaScript or C based application 
during Switch Startup to set some Global Variables based on data in an XML file.

Is there a way to do this currently?
Or is this a case where a new specialized module needs to be developed?

Regards,
Paul David Swardstrom



From brian at freeswitch.org  Mon May 24 11:24:44 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 24 May 2010 13:24:44 -0500
Subject: [Freeswitch-users] Searching mailing lists
In-Reply-To: <1229079085.100.1274717262156.JavaMail.root@srvr12.remotelinkml.com>
References: <1229079085.100.1274717262156.JavaMail.root@srvr12.remotelinkml.com>
Message-ID: <0A8EE721-1247-4E78-9B99-BEFB156799A6@freeswitch.org>

Google does a great job of searching the list...

"site:lists.freeswitch.org [searchtermshere]"

Google foo be weak with the!

/b



On May 24, 2010, at 11:07 AM, David Swardstrom wrote:

> I am new to using FreeSwitch but I have a couple of questions about things.
> But before I send in the questions, I would like to see if there is any 
> discussion either in the Development or the User list.
> However, there does not seem to be a search engine capability supplied.
> Is there a search capability?
> If so, is there a web page for this?
> 
> Regards,
> Paul David Swardstrom



From testeador01 at gmail.com  Mon May 24 11:51:36 2010
From: testeador01 at gmail.com (Milena)
Date: Mon, 24 May 2010 13:51:36 -0500
Subject: [Freeswitch-users] make a remote extension call from a
	particular gateway
In-Reply-To: <AANLkTikAB5oG3ujobJ8iv6L61JSqErAM8QK7YpUhNm64@mail.gmail.com>
References: <AANLkTikAB5oG3ujobJ8iv6L61JSqErAM8QK7YpUhNm64@mail.gmail.com>
Message-ID: <AANLkTilsjLPuerrpmVXoII61mOriHZ-i60ItXB06rm98@mail.gmail.com>

hello,
that is becuase the first extension that appears on the dialplan gets parsed
first, put the domestic1008.uk one first then the other one.


2010/5/23 Wa Halla <wahalla at gmail.com>

> I have a branch office that I want to call from a particular gateway. The
> branch is on extension 1008. With the dialplan below any extension uses
> gateway4 even extension 1008
>
> <extension name="domestic.uk">
>     <condition field="${toll_allow}" expression="domestic"/>
>     <condition field="destination_number" expression="^(\d{11})$">
>       <action application="set"
> data="effective_caller_id_number=${outbound_caller_id_number}"/>
>       <action application="set"
> data="effective_caller_id_name=${outbound_caller_id_name}"/>
>       <action application="bridge" data="sofia/gateway/gateway4/$1"/>
>     </condition>
>   </extension>
>
> <extension name="domestic1008.uk">
>     <condition field="${toll_allow}" expression="domestic"/>
>     <condition field="sip_auth_username" expression="^(1008)$"/>
>     <condition field="destination_number" expression="^(\d{11})$">
>       <action application="set"
> data="effective_caller_id_number=${outbound_caller_id_number}"/>
>       <action application="set"
> data="effective_caller_id_name=${outbound_caller_id_name}"/>
>       <action application="bridge" data="sofia/gateway/gateway1/$1"/>
>     </condition>
>   </extension>
>
> Any suggestions of what I am doing wrong. I have tried <condition
> field="sip_auth_username" expression="^(1008)$"/> with and <condition
> field="sip_auth_username" expression="^1008$"/> without the brackets so have
> elimated that.
>
> If I switch the gateways in the  domestic.uk extension, all calls go from
> gateway1. So it seems the domestic1008.uk never gets executed as true.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Mon May 24 12:20:07 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 24 May 2010 14:20:07 -0500
Subject: [Freeswitch-users] Performing system wide initialization during
	startup
In-Reply-To: <833424957.122.1274725321624.JavaMail.root@srvr12.remotelinkml.com>
References: <833424957.122.1274725321624.JavaMail.root@srvr12.remotelinkml.com>
Message-ID: <AANLkTilhoM4jexcGu_DSa75fKtGnFxCPlidit4_EmzgR@mail.gmail.com>

you could use the db_hash for runtime scoped globals or the plain old db app
for persistant data storage.


On Mon, May 24, 2010 at 1:22 PM, David Swardstrom <
dswardstrom at remotelink.com> wrote:

> I am recreating a conferencing application on FreeSwitch to replace a
> proprietary application
> originally developed using Dialogic ISDN hardware.
> I am using mod_conference and a JavaScript application based on the
> JavaScript samples.
>
> As part of this application, I have to track some counters and other data
> that are global
> to all of the conference/channel instances. I would like to initialize this
> code during
> startup. I could do this multiple ways including using C or C++ code.
> I would prefer to accomplish this by running a small JavaScript or C based
> application
> during Switch Startup to set some Global Variables based on data in an XML
> file.
>
> Is there a way to do this currently?
> Or is this a case where a new specialized module needs to be developed?
>
> Regards,
> Paul David Swardstrom
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <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
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From davis.erwin at gmail.com  Mon May 24 12:31:40 2010
From: davis.erwin at gmail.com (Erwin Davis)
Date: Mon, 24 May 2010 12:31:40 -0700
Subject: [Freeswitch-users] curl_xml
In-Reply-To: <AANLkTimR6zWJX3bwGCuvt5efDRlj3qY_Xiisn3doGupD@mail.gmail.com>
References: <AANLkTiltmx-qzAAgkQjUcCAGsAMp9_X573zbQic24pf7@mail.gmail.com>
	<AANLkTin2KFDpjFdxDIYsocYg1bH0bIQsExKnrrOtHM73@mail.gmail.com>
	<1B2B2257-94F7-47BD-938A-369FB792D2D8@freeswitch.org>
	<AANLkTinaUDJi5uXrollpVSXdgcpE7VNW79ikWcFdt07Y@mail.gmail.com>
	<AANLkTin61Y5yQWs--C1RggNo9ErnGqEYXhhSQ4XV68O1@mail.gmail.com>
	<AANLkTim6sWASDmj8w6KF9gsgB8joPKZC6VX34HpEOCdE@mail.gmail.com>
	<AANLkTimWDqSf2a7ZacI6q_4AQP5wm1sJImEKZBX3QzbC@mail.gmail.com>
	<AANLkTilAIOnMu1bmJBOTsCvrnSfytRE8tenp6JEVTMc-@mail.gmail.com>
	<AANLkTin9OKJNURCKZ7__EMGyo1_4WYXufdQhDmYAcJOb@mail.gmail.com>
	<AANLkTimR6zWJX3bwGCuvt5efDRlj3qY_Xiisn3doGupD@mail.gmail.com>
Message-ID: <AANLkTinsj5G5R2fCYFmDyhEmIu4empwzE0psrUU-q6fL@mail.gmail.com>

Hi, Vitalli,

Thanks for your PHP code. Unfortunately, I can't run that code because my
project is in rails.
But I didnot see the difference between you code and mine. What does the xml
log file look like when you turn on "xml_curl debug_on"? Could you cut and
paste them into  freeswitch pastebin?

My code is cloned from
http://wiki.freeswitch.org/wiki/Mod_xml_curl_Ruby_directory_example. I am
not sure that the xml output may cause some problem.

When I type "find_use_xml  id 1018 192.168.1.31", the FS ignored the
response from web server. Instead, it retrieve the info from the local
directory.

freeswitch at proxy1.voiceserver.com> find_user_xml id 1018  192.168.1.31
2010-05-24 15:12:12.239449 [CONSOLE] mod_xml_curl.c:299 XML response is in
/usr/local/freeswitch/
scripts803ca405-c463-4deb-8121-982e1251831a.tmp.xml
API CALL [find_user_xml(id 1018  192.168.1.31)] output:
<user id="1018">
      <params>
          <param name="password" value="1234"></param>   <<<<===== the
server response with "1018", "1234" is the local default setting

          <param name="vm-password" value="1018"></param>

    </params>

      <variables>
          <variable name="toll_allow"
value="domestic,international,local"></variable>

          <variable name="accountcode" value="1018"></variable>

          <variable name="user_context" value="default"></variable>

          <variable name="effective_caller_id_name" value="Extension
1018"></variable>

          <variable name="effective_caller_id_number"
value="1018"></variable>

          <variable name="outbound_caller_id_name" value="MyPBX"></variable>

          <variable name="outbound_caller_id_number"
value="0000000000"></variable>

          <variable name="callgroup" value="techsupport"></variable>

    </variables>

  </user>

Below is the rails code to output the xml response.
==============================================
========

  def directory

    if params['user'].nil? || params['user'].empty?
      ##if not looking for a user info, send 404 Not Found
      @out_xml = Output404NotFound()
    else
      @username = params['user'].to_s
      puts "username = " + @username
      @domain = params['domain'].to_s
      puts params
      @dir=Directory.find_by_effective_caller_id_number(@username)
      puts @dir
      if @dir.nil?
        @out_xml = Output404NotFound()
      else
        @out_xml = OutputXMLResponse()
      end
    end
    puts "reply for directory query: " + @out_xml + "\n"
  end

private

  def Output404NotFound
    xml_text = "<?xml version=\"1.0\" encoding=\"UTF-8\" standalone=\"no\"?>
                 <document type=\"freeswitch/xml\">
                     <section name=\"result\">
                          <result status=\"not found\" />
                     </section>
                 </document> "
    return xml_text
  end

  def OutputXMLResponse
    xml_text =
        "<?xml version=\"1.0\" encoding=\"UTF-8\" standalone=\"no\"?>
           <document type=\"freeswitch/xml\">
             <section name=\"directory\" description=\"User Directory\">
                 <domain name=\"#{@domain}\">
                      <params>
                         <param name=\"dial-string\"
value=\"{presence_id=${dialed_user}@
${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}\"/>
                      </params>

                       <groups>
                           <group name=\"default\">
                               <users>
                                  <user id=\"#{@username}\">
                                     <params>
                                        <param name=\"password\" value=\"" +
@dir.password + "\"/>
                                     </params>
                                  </user>
                               </users>
                           </group>
                         </groups>
                   </domain>
             </section>
         </document> "
     puts "reply for directory query: " + xml_text + "\n"
     return xml_text
  end



On Mon, May 24, 2010 at 8:38 AM, Vitalii Colosov <vetali100 at gmail.com>wrote:

> Just to avoid any misunderstanding - this is to check all other things
> around freeswitch configuration.
>
> If this will work, it will be very easy to add database lookup for the user
> trying to register:
>
>
> $user = $_POST['user'];
> $query="select password from your_users_table where user ='$user'";
> $result=mysqli_query($db_connection, $query);
>
> ...get user and password from the result
>
> And then use $user and $password variables instead of hard-coded "1018",
> "1018" in the code I provided.
>
> Really hope this helps, it works for me.
>
> Regards,
> Vitalie
>
>
> 2010/5/24 Vitalii Colosov <vetali100 at gmail.com>
>
>> Sorry there was an else related to the missing entry, i removed it.
>>
>> Please try another one.
>>
>> 2010/5/24 Vitalii Colosov <vetali100 at gmail.com>
>>
>> Please try to use my attached php script, if you can.
>>>
>>> I changed it to fit your data (user 1018, pass: 1018), but I did not
>>> check for any syntax error (hope it does not have :-) ).
>>>
>>> Please reply back on result.
>>>
>>> Regards,
>>>
>>> Vitalie
>>>
>>>
>>> 2010/5/24 Erwin Davis <davis.erwin at gmail.com>
>>>
>>>> yeah, my code replied with line 29-46. I donot know where the rest of
>>>> html code was generated.
>>>>
>>>> On Mon, May 24, 2010 at 10:34 AM, Milena <testeador01 at gmail.com> wrote:
>>>>
>>>>> Hello,
>>>>> You're not meant to send a whole website full of html tags as a
>>>>> response.
>>>>> Just reply the part of the lines 29 - 46 on your pastebin and get rid
>>>>> of the rest.
>>>>>
>>>>>
>>>>> 2010,
>>>>> /5/24 Erwin Davis <davis.erwin at gmail.com>
>>>>>
>>>>> Hi, I am stuck, My console log is in
>>>>>> http://pastebin.freeswitch.org/13030 and my xml log is in
>>>>>> http://pastebin.freeswitch.org/13029.  I guess that the problem
>>>>>> should be pretty small but I just can not figure it out. -:(  Thanks for
>>>>>> your help.
>>>>>>
>>>>>> On Mon, May 24, 2010 at 8:12 AM, Vitalii Colosov <vetali100 at gmail.com
>>>>>> > wrote:
>>>>>>
>>>>>>> Hi Erwin,
>>>>>>> Maybe you need to add the first line <?xml version="1.0"
>>>>>>> encoding="UTF-8" standalone="no"?>
>>>>>>>
>>>>>>> This is what I have and it works perfectly, tested right now:
>>>>>>>
>>>>>>> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>>>>>>> <document type="freeswitch/xml">
>>>>>>>   <section name="directory">
>>>>>>>     <domain name="$${domain}">
>>>>>>>       <params>
>>>>>>>         <param name="dial-string" value="{presence_id=${dialed_user}@
>>>>>>> ${dialed_domain}}${sofia_contact(${dialed_user}@
>>>>>>> ${dialed_domain})}"/>
>>>>>>>       </params>
>>>>>>>       <groups>
>>>>>>>         <group name="default">
>>>>>>>           <users>
>>>>>>>             <user id="1000">
>>>>>>>               <params>
>>>>>>>                 <param name="a1-hash" value="somehashvalue"/>
>>>>>>>                 <param name="vm-a1-hash" value="somehashvalue"/>
>>>>>>>               </params>
>>>>>>>               <variables>
>>>>>>>                 <variable name="user_context" value="default"/>
>>>>>>>                 <variable name="accountcode" value="1000"/>
>>>>>>>               </variables>
>>>>>>>             </user>
>>>>>>>           </users>
>>>>>>>         </group>
>>>>>>>       </groups>
>>>>>>>     </domain>
>>>>>>>   </section>
>>>>>>> </document>
>>>>>>>
>>>>>>> Regards,
>>>>>>> Vitalie
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From vetali100 at gmail.com  Mon May 24 12:45:59 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Mon, 24 May 2010 22:45:59 +0300
Subject: [Freeswitch-users] curl_xml
In-Reply-To: <AANLkTinsj5G5R2fCYFmDyhEmIu4empwzE0psrUU-q6fL@mail.gmail.com>
References: <AANLkTiltmx-qzAAgkQjUcCAGsAMp9_X573zbQic24pf7@mail.gmail.com>
	<1B2B2257-94F7-47BD-938A-369FB792D2D8@freeswitch.org>
	<AANLkTinaUDJi5uXrollpVSXdgcpE7VNW79ikWcFdt07Y@mail.gmail.com>
	<AANLkTin61Y5yQWs--C1RggNo9ErnGqEYXhhSQ4XV68O1@mail.gmail.com>
	<AANLkTim6sWASDmj8w6KF9gsgB8joPKZC6VX34HpEOCdE@mail.gmail.com>
	<AANLkTimWDqSf2a7ZacI6q_4AQP5wm1sJImEKZBX3QzbC@mail.gmail.com>
	<AANLkTilAIOnMu1bmJBOTsCvrnSfytRE8tenp6JEVTMc-@mail.gmail.com>
	<AANLkTin9OKJNURCKZ7__EMGyo1_4WYXufdQhDmYAcJOb@mail.gmail.com>
	<AANLkTimR6zWJX3bwGCuvt5efDRlj3qY_Xiisn3doGupD@mail.gmail.com>
	<AANLkTinsj5G5R2fCYFmDyhEmIu4empwzE0psrUU-q6fL@mail.gmail.com>
Message-ID: <AANLkTilU7uEEjVhVVo6OAObi7Bq9o2cAmC1rmhrlglr3@mail.gmail.com>

Sorry, I never used rubby, no help from my side here...

This is my XML response, it's not so long to put it in pastebin, also I
never used pastebin yet:

<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
  <section name="directory">
    <domain name="$${domain}">
      <params>
        <param name="dial-string" value="{presence_id=${dialed_user}@
${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
      </params>
      <groups>
        <group name="default">
          <users>
            <user id="1000">
              <params>
                <param name="a1-hash" value="somehashvalue"/>
                <param name="vm-a1-hash" value="somehashvalue"/>
              </params>
              <variables>
                <variable name="user_context" value="default"/>
                <variable name="accountcode" value="1000"/>
              </variables>
            </user>
          </users>
        </group>
      </groups>
    </domain>
  </section>
</document>

Regards,
Vitalie

2010/5/24 Erwin Davis <davis.erwin at gmail.com>

> Hi, Vitalli,
>
> Thanks for your PHP code. Unfortunately, I can't run that code because my
> project is in rails.
> But I didnot see the difference between you code and mine. What does the
> xml log file look like when you turn on "xml_curl debug_on"? Could you cut
> and paste them into  freeswitch pastebin?
>
> My code is cloned from
> http://wiki.freeswitch.org/wiki/Mod_xml_curl_Ruby_directory_example. I am
> not sure that the xml output may cause some problem.
>
> When I type "find_use_xml  id 1018 192.168.1.31", the FS ignored the
> response from web server. Instead, it retrieve the info from the local
> directory.
>
> freeswitch at proxy1.voiceserver.com> find_user_xml id 1018  192.168.1.31
> 2010-05-24 15:12:12.239449 [CONSOLE] mod_xml_curl.c:299 XML response is in
> /usr/local/freeswitch/
> scripts803ca405-c463-4deb-8121-982e1251831a.tmp.xml
> API CALL [find_user_xml(id 1018  192.168.1.31)] output:
> <user id="1018">
>       <params>
>           <param name="password" value="1234"></param>   <<<<===== the
> server response with "1018", "1234" is the local default setting
>
>           <param name="vm-password" value="1018"></param>
>
>     </params>
>
>       <variables>
>           <variable name="toll_allow"
> value="domestic,international,local"></variable>
>
>           <variable name="accountcode" value="1018"></variable>
>
>
>           <variable name="user_context" value="default"></variable>
>
>           <variable name="effective_caller_id_name" value="Extension
> 1018"></variable>
>
>           <variable name="effective_caller_id_number"
> value="1018"></variable>
>
>           <variable name="outbound_caller_id_name"
> value="MyPBX"></variable>
>
>           <variable name="outbound_caller_id_number"
> value="0000000000"></variable>
>
>           <variable name="callgroup" value="techsupport"></variable>
>
>     </variables>
>
>   </user>
>
> Below is the rails code to output the xml response.
> ==============================================
> ========
>
>   def directory
>
>     if params['user'].nil? || params['user'].empty?
>       ##if not looking for a user info, send 404 Not Found
>       @out_xml = Output404NotFound()
>     else
>       @username = params['user'].to_s
>       puts "username = " + @username
>       @domain = params['domain'].to_s
>       puts params
>       @dir=Directory.find_by_effective_caller_id_number(@username)
>       puts @dir
>       if @dir.nil?
>         @out_xml = Output404NotFound()
>       else
>         @out_xml = OutputXMLResponse()
>       end
>     end
>     puts "reply for directory query: " + @out_xml + "\n"
>   end
>
> private
>
>   def Output404NotFound
>     xml_text = "<?xml version=\"1.0\" encoding=\"UTF-8\"
> standalone=\"no\"?>
>                  <document type=\"freeswitch/xml\">
>                      <section name=\"result\">
>                           <result status=\"not found\" />
>                      </section>
>                  </document> "
>     return xml_text
>   end
>
>   def OutputXMLResponse
>     xml_text =
>         "<?xml version=\"1.0\" encoding=\"UTF-8\" standalone=\"no\"?>
>            <document type=\"freeswitch/xml\">
>              <section name=\"directory\" description=\"User Directory\">
>
>                  <domain name=\"#{@domain}\">
>                       <params>
>                          <param name=\"dial-string\"
> value=\"{presence_id=${dialed_user}@
> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}\"/>
>                       </params>
>
>                        <groups>
>                            <group name=\"default\">
>                                <users>
>                                   <user id=\"#{@username}\">
>                                      <params>
>                                         <param name=\"password\" value=\""
> + @dir.password + "\"/>
>                                      </params>
>
>                                   </user>
>                                </users>
>                            </group>
>                          </groups>
>                    </domain>
>              </section>
>          </document> "
>      puts "reply for directory query: " + xml_text + "\n"
>      return xml_text
>   end
>
>
>
>
> On Mon, May 24, 2010 at 8:38 AM, Vitalii Colosov <vetali100 at gmail.com>wrote:
>
>> Just to avoid any misunderstanding - this is to check all other things
>> around freeswitch configuration.
>>
>> If this will work, it will be very easy to add database lookup for the
>> user trying to register:
>>
>>
>> $user = $_POST['user'];
>> $query="select password from your_users_table where user ='$user'";
>> $result=mysqli_query($db_connection, $query);
>>
>> ...get user and password from the result
>>
>> And then use $user and $password variables instead of hard-coded "1018",
>> "1018" in the code I provided.
>>
>> Really hope this helps, it works for me.
>>
>> Regards,
>> Vitalie
>>
>>
>> 2010/5/24 Vitalii Colosov <vetali100 at gmail.com>
>>
>>> Sorry there was an else related to the missing entry, i removed it.
>>>
>>> Please try another one.
>>>
>>> 2010/5/24 Vitalii Colosov <vetali100 at gmail.com>
>>>
>>> Please try to use my attached php script, if you can.
>>>>
>>>> I changed it to fit your data (user 1018, pass: 1018), but I did not
>>>> check for any syntax error (hope it does not have :-) ).
>>>>
>>>> Please reply back on result.
>>>>
>>>> Regards,
>>>>
>>>> Vitalie
>>>>
>>>>
>>>> 2010/5/24 Erwin Davis <davis.erwin at gmail.com>
>>>>
>>>>> yeah, my code replied with line 29-46. I donot know where the rest of
>>>>> html code was generated.
>>>>>
>>>>> On Mon, May 24, 2010 at 10:34 AM, Milena <testeador01 at gmail.com>wrote:
>>>>>
>>>>>> Hello,
>>>>>> You're not meant to send a whole website full of html tags as a
>>>>>> response.
>>>>>> Just reply the part of the lines 29 - 46 on your pastebin and get rid
>>>>>> of the rest.
>>>>>>
>>>>>>
>>>>>> 2010,
>>>>>> /5/24 Erwin Davis <davis.erwin at gmail.com>
>>>>>>
>>>>>> Hi, I am stuck, My console log is in
>>>>>>> http://pastebin.freeswitch.org/13030 and my xml log is in
>>>>>>> http://pastebin.freeswitch.org/13029.  I guess that the problem
>>>>>>> should be pretty small but I just can not figure it out. -:(  Thanks for
>>>>>>> your help.
>>>>>>>
>>>>>>> On Mon, May 24, 2010 at 8:12 AM, Vitalii Colosov <
>>>>>>> vetali100 at gmail.com> wrote:
>>>>>>>
>>>>>>>> Hi Erwin,
>>>>>>>> Maybe you need to add the first line <?xml version="1.0"
>>>>>>>> encoding="UTF-8" standalone="no"?>
>>>>>>>>
>>>>>>>> This is what I have and it works perfectly, tested right now:
>>>>>>>>
>>>>>>>> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>>>>>>>> <document type="freeswitch/xml">
>>>>>>>>   <section name="directory">
>>>>>>>>     <domain name="$${domain}">
>>>>>>>>       <params>
>>>>>>>>         <param name="dial-string"
>>>>>>>> value="{presence_id=${dialed_user}@
>>>>>>>> ${dialed_domain}}${sofia_contact(${dialed_user}@
>>>>>>>> ${dialed_domain})}"/>
>>>>>>>>       </params>
>>>>>>>>       <groups>
>>>>>>>>         <group name="default">
>>>>>>>>           <users>
>>>>>>>>             <user id="1000">
>>>>>>>>               <params>
>>>>>>>>                 <param name="a1-hash" value="somehashvalue"/>
>>>>>>>>                 <param name="vm-a1-hash" value="somehashvalue"/>
>>>>>>>>               </params>
>>>>>>>>               <variables>
>>>>>>>>                 <variable name="user_context" value="default"/>
>>>>>>>>                 <variable name="accountcode" value="1000"/>
>>>>>>>>               </variables>
>>>>>>>>             </user>
>>>>>>>>           </users>
>>>>>>>>         </group>
>>>>>>>>       </groups>
>>>>>>>>     </domain>
>>>>>>>>   </section>
>>>>>>>> </document>
>>>>>>>>
>>>>>>>> Regards,
>>>>>>>> Vitalie
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ranjtech at gmail.com  Mon May 24 13:45:01 2010
From: ranjtech at gmail.com (RR)
Date: Mon, 24 May 2010 16:45:01 -0400
Subject: [Freeswitch-users] Question about inbound ACL
Message-ID: <AANLkTimAepVJa1kK9Qfut7jRZK1q4zjIjzUfsvu18nZt@mail.gmail.com>

Hello List,

I have tried to read as much as I could and try out a bunch of things but it
doesn't seem to be working. The scenario is that we have a FS setup as a
call distributor to our internal SIP servers. The calls come into the FS
when people call any of the 6000 or so DIDs we own. These DIDs are through
various providers who's IP addresses we know (obviously). I want to receive
/ process only calls from these external IP addresses and nothing else. So I
did the following in acl.conf.xml file

        <list name="DIDProviders" default="deny">
          <node type="allow" cidr="64.XXX.XXX.XXX/32"/>
          <node type="allow" cidr="67.XXX.XXX.XXX/30"/>
           etc...
        </list>

then I went into the $FSHOME/conf/sip_profiles/internal.xml and did the
following:

<param name="apply-inbound-acl" value="DIDProviders"/>

but I still keep seeing calls from other IPs that are not in the
"DIDProviders" list getting through.

What else do I need to do to prevent this? Should this not be in
internal.xml but in external.xml?

TIA
\RR
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From davis.erwin at gmail.com  Mon May 24 13:47:57 2010
From: davis.erwin at gmail.com (Erwin Davis)
Date: Mon, 24 May 2010 16:47:57 -0400
Subject: [Freeswitch-users] curl_xml
In-Reply-To: <AANLkTilU7uEEjVhVVo6OAObi7Bq9o2cAmC1rmhrlglr3@mail.gmail.com>
References: <AANLkTiltmx-qzAAgkQjUcCAGsAMp9_X573zbQic24pf7@mail.gmail.com>
	<AANLkTinaUDJi5uXrollpVSXdgcpE7VNW79ikWcFdt07Y@mail.gmail.com>
	<AANLkTin61Y5yQWs--C1RggNo9ErnGqEYXhhSQ4XV68O1@mail.gmail.com>
	<AANLkTim6sWASDmj8w6KF9gsgB8joPKZC6VX34HpEOCdE@mail.gmail.com>
	<AANLkTimWDqSf2a7ZacI6q_4AQP5wm1sJImEKZBX3QzbC@mail.gmail.com>
	<AANLkTilAIOnMu1bmJBOTsCvrnSfytRE8tenp6JEVTMc-@mail.gmail.com>
	<AANLkTin9OKJNURCKZ7__EMGyo1_4WYXufdQhDmYAcJOb@mail.gmail.com>
	<AANLkTimR6zWJX3bwGCuvt5efDRlj3qY_Xiisn3doGupD@mail.gmail.com>
	<AANLkTinsj5G5R2fCYFmDyhEmIu4empwzE0psrUU-q6fL@mail.gmail.com>
	<AANLkTilU7uEEjVhVVo6OAObi7Bq9o2cAmC1rmhrlglr3@mail.gmail.com>
Message-ID: <AANLkTildFYq1rtaLicpS7kFnrRWk5W1GxQNMyFJMy9tL@mail.gmail.com>

Hello,
 Can anyone post a xml log file when you turn on "xml_curl debug_on"? I hope
to compare with the my log file with a valid one? Thanks,

Regards,

e

On Mon, May 24, 2010 at 3:45 PM, Vitalii Colosov <vetali100 at gmail.com>wrote:

> Sorry, I never used rubby, no help from my side here...
>
> This is my XML response, it's not so long to put it in pastebin, also I
> never used pastebin yet:
>
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
>   <section name="directory">
>     <domain name="$${domain}">
>       <params>
>         <param name="dial-string" value="{presence_id=${dialed_user}@
> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>       </params>
>       <groups>
>         <group name="default">
>           <users>
>             <user id="1000">
>               <params>
>                 <param name="a1-hash" value="somehashvalue"/>
>                 <param name="vm-a1-hash" value="somehashvalue"/>
>               </params>
>               <variables>
>                 <variable name="user_context" value="default"/>
>                 <variable name="accountcode" value="1000"/>
>               </variables>
>             </user>
>           </users>
>         </group>
>       </groups>
>     </domain>
>   </section>
> </document>
>
> Regards,
> Vitalie
>
> 2010/5/24 Erwin Davis <davis.erwin at gmail.com>
>
>> Hi, Vitalli,
>>
>> Thanks for your PHP code. Unfortunately, I can't run that code because my
>> project is in rails.
>> But I didnot see the difference between you code and mine. What does the
>> xml log file look like when you turn on "xml_curl debug_on"? Could you cut
>> and paste them into  freeswitch pastebin?
>>
>> My code is cloned from
>> http://wiki.freeswitch.org/wiki/Mod_xml_curl_Ruby_directory_example. I am
>> not sure that the xml output may cause some problem.
>>
>> When I type "find_use_xml  id 1018 192.168.1.31", the FS ignored the
>> response from web server. Instead, it retrieve the info from the local
>> directory.
>>
>> freeswitch at proxy1.voiceserver.com> find_user_xml id 1018  192.168.1.31
>> 2010-05-24 15:12:12.239449 [CONSOLE] mod_xml_curl.c:299 XML response is in
>> /usr/local/freeswitch/
>> scripts803ca405-c463-4deb-8121-982e1251831a.tmp.xml
>> API CALL [find_user_xml(id 1018  192.168.1.31)] output:
>> <user id="1018">
>>       <params>
>>           <param name="password" value="1234"></param>   <<<<===== the
>> server response with "1018", "1234" is the local default setting
>>
>>           <param name="vm-password" value="1018"></param>
>>
>>     </params>
>>
>>       <variables>
>>           <variable name="toll_allow"
>> value="domestic,international,local"></variable>
>>
>>           <variable name="accountcode" value="1018"></variable>
>>
>>
>>           <variable name="user_context" value="default"></variable>
>>
>>           <variable name="effective_caller_id_name" value="Extension
>> 1018"></variable>
>>
>>           <variable name="effective_caller_id_number"
>> value="1018"></variable>
>>
>>           <variable name="outbound_caller_id_name"
>> value="MyPBX"></variable>
>>
>>           <variable name="outbound_caller_id_number"
>> value="0000000000"></variable>
>>
>>           <variable name="callgroup" value="techsupport"></variable>
>>
>>     </variables>
>>
>>   </user>
>>
>> Below is the rails code to output the xml response.
>> ==============================================
>> ========
>>
>>   def directory
>>
>>     if params['user'].nil? || params['user'].empty?
>>       ##if not looking for a user info, send 404 Not Found
>>       @out_xml = Output404NotFound()
>>     else
>>       @username = params['user'].to_s
>>       puts "username = " + @username
>>       @domain = params['domain'].to_s
>>       puts params
>>       @dir=Directory.find_by_effective_caller_id_number(@username)
>>       puts @dir
>>       if @dir.nil?
>>         @out_xml = Output404NotFound()
>>       else
>>         @out_xml = OutputXMLResponse()
>>       end
>>     end
>>     puts "reply for directory query: " + @out_xml + "\n"
>>   end
>>
>> private
>>
>>   def Output404NotFound
>>     xml_text = "<?xml version=\"1.0\" encoding=\"UTF-8\"
>> standalone=\"no\"?>
>>                  <document type=\"freeswitch/xml\">
>>                      <section name=\"result\">
>>                           <result status=\"not found\" />
>>                      </section>
>>                  </document> "
>>     return xml_text
>>   end
>>
>>   def OutputXMLResponse
>>     xml_text =
>>         "<?xml version=\"1.0\" encoding=\"UTF-8\" standalone=\"no\"?>
>>            <document type=\"freeswitch/xml\">
>>              <section name=\"directory\" description=\"User Directory\">
>>
>>                  <domain name=\"#{@domain}\">
>>                       <params>
>>                          <param name=\"dial-string\"
>> value=\"{presence_id=${dialed_user}@
>> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}\"/>
>>                       </params>
>>
>>                        <groups>
>>                            <group name=\"default\">
>>                                <users>
>>                                    <user id=\"#{@username}\">
>>                                      <params>
>>                                         <param name=\"password\" value=\""
>> + @dir.password + "\"/>
>>                                      </params>
>>
>>                                   </user>
>>                                </users>
>>                            </group>
>>                          </groups>
>>                    </domain>
>>              </section>
>>          </document> "
>>      puts "reply for directory query: " + xml_text + "\n"
>>      return xml_text
>>   end
>>
>>
>>
>>
>> On Mon, May 24, 2010 at 8:38 AM, Vitalii Colosov <vetali100 at gmail.com>wrote:
>>
>>> Just to avoid any misunderstanding - this is to check all other things
>>> around freeswitch configuration.
>>>
>>> If this will work, it will be very easy to add database lookup for the
>>> user trying to register:
>>>
>>>
>>> $user = $_POST['user'];
>>> $query="select password from your_users_table where user ='$user'";
>>> $result=mysqli_query($db_connection, $query);
>>>
>>> ...get user and password from the result
>>>
>>> And then use $user and $password variables instead of hard-coded "1018",
>>> "1018" in the code I provided.
>>>
>>> Really hope this helps, it works for me.
>>>
>>> Regards,
>>> Vitalie
>>>
>>>
>>> 2010/5/24 Vitalii Colosov <vetali100 at gmail.com>
>>>
>>>> Sorry there was an else related to the missing entry, i removed it.
>>>>
>>>> Please try another one.
>>>>
>>>> 2010/5/24 Vitalii Colosov <vetali100 at gmail.com>
>>>>
>>>> Please try to use my attached php script, if you can.
>>>>>
>>>>> I changed it to fit your data (user 1018, pass: 1018), but I did not
>>>>> check for any syntax error (hope it does not have :-) ).
>>>>>
>>>>> Please reply back on result.
>>>>>
>>>>> Regards,
>>>>>
>>>>> Vitalie
>>>>>
>>>>>
>>>>> 2010/5/24 Erwin Davis <davis.erwin at gmail.com>
>>>>>
>>>>>> yeah, my code replied with line 29-46. I donot know where the rest of
>>>>>> html code was generated.
>>>>>>
>>>>>> On Mon, May 24, 2010 at 10:34 AM, Milena <testeador01 at gmail.com>wrote:
>>>>>>
>>>>>>> Hello,
>>>>>>> You're not meant to send a whole website full of html tags as a
>>>>>>> response.
>>>>>>> Just reply the part of the lines 29 - 46 on your pastebin and get rid
>>>>>>> of the rest.
>>>>>>>
>>>>>>>
>>>>>>> 2010,
>>>>>>> /5/24 Erwin Davis <davis.erwin at gmail.com>
>>>>>>>
>>>>>>> Hi, I am stuck, My console log is in
>>>>>>>> http://pastebin.freeswitch.org/13030 and my xml log is in
>>>>>>>> http://pastebin.freeswitch.org/13029.  I guess that the problem
>>>>>>>> should be pretty small but I just can not figure it out. -:(  Thanks for
>>>>>>>> your help.
>>>>>>>>
>>>>>>>> On Mon, May 24, 2010 at 8:12 AM, Vitalii Colosov <
>>>>>>>> vetali100 at gmail.com> wrote:
>>>>>>>>
>>>>>>>>> Hi Erwin,
>>>>>>>>> Maybe you need to add the first line <?xml version="1.0"
>>>>>>>>> encoding="UTF-8" standalone="no"?>
>>>>>>>>>
>>>>>>>>> This is what I have and it works perfectly, tested right now:
>>>>>>>>>
>>>>>>>>> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>>>>>>>>> <document type="freeswitch/xml">
>>>>>>>>>   <section name="directory">
>>>>>>>>>     <domain name="$${domain}">
>>>>>>>>>       <params>
>>>>>>>>>         <param name="dial-string"
>>>>>>>>> value="{presence_id=${dialed_user}@
>>>>>>>>> ${dialed_domain}}${sofia_contact(${dialed_user}@
>>>>>>>>> ${dialed_domain})}"/>
>>>>>>>>>       </params>
>>>>>>>>>       <groups>
>>>>>>>>>         <group name="default">
>>>>>>>>>           <users>
>>>>>>>>>             <user id="1000">
>>>>>>>>>               <params>
>>>>>>>>>                 <param name="a1-hash" value="somehashvalue"/>
>>>>>>>>>                 <param name="vm-a1-hash" value="somehashvalue"/>
>>>>>>>>>               </params>
>>>>>>>>>               <variables>
>>>>>>>>>                 <variable name="user_context" value="default"/>
>>>>>>>>>                 <variable name="accountcode" value="1000"/>
>>>>>>>>>               </variables>
>>>>>>>>>             </user>
>>>>>>>>>           </users>
>>>>>>>>>         </group>
>>>>>>>>>       </groups>
>>>>>>>>>     </domain>
>>>>>>>>>   </section>
>>>>>>>>> </document>
>>>>>>>>>
>>>>>>>>> Regards,
>>>>>>>>> Vitalie
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at gmail.com  Mon May 24 15:26:51 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Tue, 25 May 2010 00:26:51 +0200
Subject: [Freeswitch-users] Question about inbound ACL
In-Reply-To: <AANLkTimAepVJa1kK9Qfut7jRZK1q4zjIjzUfsvu18nZt@mail.gmail.com>
References: <AANLkTimAepVJa1kK9Qfut7jRZK1q4zjIjzUfsvu18nZt@mail.gmail.com>
Message-ID: <5ED93CC6-A71B-417B-8B5B-565193F2D9B5@gmail.com>

Well you need to put the ACL in the SIP Profile your providers are  
sending calls to.
By default, it's port 5080 for external and port 5060 for internal.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 24/05/2010 ? 22:45, RR a ?crit :

> Hello List,
>
> I have tried to read as much as I could and try out a bunch of  
> things but it doesn't seem to be working. The scenario is that we  
> have a FS setup as a call distributor to our internal SIP servers.  
> The calls come into the FS when people call any of the 6000 or so  
> DIDs we own. These DIDs are through various providers who's IP  
> addresses we know (obviously). I want to receive / process only  
> calls from these external IP addresses and nothing else. So I did  
> the following in acl.conf.xml file
>
>         <list name="DIDProviders" default="deny">
>           <node type="allow" cidr="64.XXX.XXX.XXX/32"/>
>           <node type="allow" cidr="67.XXX.XXX.XXX/30"/>
>            etc...
>         </list>
>
> then I went into the $FSHOME/conf/sip_profiles/internal.xml and did  
> the following:
>
> <param name="apply-inbound-acl" value="DIDProviders"/>
>
> but I still keep seeing calls from other IPs that are not in the  
> "DIDProviders" list getting through.
>
> What else do I need to do to prevent this? Should this not be in  
> internal.xml but in external.xml?
>
> TIA
> \RR
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From ranjtech at gmail.com  Mon May 24 15:56:14 2010
From: ranjtech at gmail.com (RR)
Date: Mon, 24 May 2010 18:56:14 -0400
Subject: [Freeswitch-users] Question about inbound ACL
In-Reply-To: <5ED93CC6-A71B-417B-8B5B-565193F2D9B5@gmail.com>
References: <AANLkTimAepVJa1kK9Qfut7jRZK1q4zjIjzUfsvu18nZt@mail.gmail.com>
	<5ED93CC6-A71B-417B-8B5B-565193F2D9B5@gmail.com>
Message-ID: <AANLkTilp6klOIYWpa3GT3MyQYF7foJYt6zM1Tt9-H7G7@mail.gmail.com>

Hi David,

the calls are coming in on port 5060, so I'm assuming it's triggering the
internal profile(?) and as mentioned in my email, we have the acl applied in
the internal.xml file. Yet we see calls from all sorts of IPs. Anything else
we need to look at?

Thanks
RR

On Mon, May 24, 2010 at 6:26 PM, David Ponzone <david.ponzone at gmail.com>wrote:

> Well you need to put the ACL in the SIP Profile your providers are sending
> calls to.
> By default, it's port 5080 for external and port 5060 for internal.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 24/05/2010 ? 22:45, RR a ?crit :
>
> Hello List,
>
> I have tried to read as much as I could and try out a bunch of things but
> it doesn't seem to be working. The scenario is that we have a FS setup as a
> call distributor to our internal SIP servers. The calls come into the FS
> when people call any of the 6000 or so DIDs we own. These DIDs are through
> various providers who's IP addresses we know (obviously). I want to receive
> / process only calls from these external IP addresses and nothing else. So I
> did the following in acl.conf.xml file
>
>         <list name="DIDProviders" default="deny">
>           <node type="allow" cidr="64.XXX.XXX.XXX/32"/>
>           <node type="allow" cidr="67.XXX.XXX.XXX/30"/>
>            etc...
>         </list>
>
> then I went into the $FSHOME/conf/sip_profiles/internal.xml and did the
> following:
>
> <param name="apply-inbound-acl" value="DIDProviders"/>
>
> but I still keep seeing calls from other IPs that are not in the
> "DIDProviders" list getting through.
>
> What else do I need to do to prevent this? Should this not be in
> internal.xml but in external.xml?
>
> TIA
> \RR
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Mon May 24 16:36:48 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 24 May 2010 18:36:48 -0500
Subject: [Freeswitch-users] curl_xml
In-Reply-To: <AANLkTildFYq1rtaLicpS7kFnrRWk5W1GxQNMyFJMy9tL@mail.gmail.com>
References: <AANLkTiltmx-qzAAgkQjUcCAGsAMp9_X573zbQic24pf7@mail.gmail.com>
	<AANLkTin61Y5yQWs--C1RggNo9ErnGqEYXhhSQ4XV68O1@mail.gmail.com>
	<AANLkTim6sWASDmj8w6KF9gsgB8joPKZC6VX34HpEOCdE@mail.gmail.com>
	<AANLkTimWDqSf2a7ZacI6q_4AQP5wm1sJImEKZBX3QzbC@mail.gmail.com>
	<AANLkTilAIOnMu1bmJBOTsCvrnSfytRE8tenp6JEVTMc-@mail.gmail.com>
	<AANLkTin9OKJNURCKZ7__EMGyo1_4WYXufdQhDmYAcJOb@mail.gmail.com>
	<AANLkTimR6zWJX3bwGCuvt5efDRlj3qY_Xiisn3doGupD@mail.gmail.com>
	<AANLkTinsj5G5R2fCYFmDyhEmIu4empwzE0psrUU-q6fL@mail.gmail.com>
	<AANLkTilU7uEEjVhVVo6OAObi7Bq9o2cAmC1rmhrlglr3@mail.gmail.com>
	<AANLkTildFYq1rtaLicpS7kFnrRWk5W1GxQNMyFJMy9tL@mail.gmail.com>
Message-ID: <AANLkTimHilHi_YnSQj58IjUHCPxcwxf4wcNfhXlOSmuI@mail.gmail.com>

I am jumping in here because I am concerned that my browser will crash if
this thread gets any bigger:

Perhaps if you change your point of view you will find it faster.
You have multiple accounts from other users that it works, but i get the
feeling you believe in the back of your mind that it's broken somehow.  If
you change your mindset to "I know I am doing something wrong and all I need
to do is figure out what it is", probably you will be able to spot the
mistake easier.

Thank you everyone who has helped here.




On Mon, May 24, 2010 at 3:47 PM, Erwin Davis <davis.erwin at gmail.com> wrote:

> Hello,
>  Can anyone post a xml log file when you turn on "xml_curl debug_on"? I
> hope to compare with the my log file with a valid one? Thanks,
>
> Regards,
>
> e
>
>
> On Mon, May 24, 2010 at 3:45 PM, Vitalii Colosov <vetali100 at gmail.com>wrote:
>
>> Sorry, I never used rubby, no help from my side here...
>>
>> This is my XML response, it's not so long to put it in pastebin, also I
>> never used pastebin yet:
>>
>> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>> <document type="freeswitch/xml">
>>   <section name="directory">
>>     <domain name="$${domain}">
>>       <params>
>>         <param name="dial-string" value="{presence_id=${dialed_user}@
>> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>>       </params>
>>       <groups>
>>         <group name="default">
>>           <users>
>>             <user id="1000">
>>               <params>
>>                 <param name="a1-hash" value="somehashvalue"/>
>>                 <param name="vm-a1-hash" value="somehashvalue"/>
>>               </params>
>>               <variables>
>>                 <variable name="user_context" value="default"/>
>>                 <variable name="accountcode" value="1000"/>
>>               </variables>
>>             </user>
>>           </users>
>>         </group>
>>       </groups>
>>     </domain>
>>   </section>
>> </document>
>>
>> Regards,
>> Vitalie
>>
>> 2010/5/24 Erwin Davis <davis.erwin at gmail.com>
>>
>>> Hi, Vitalli,
>>>
>>> Thanks for your PHP code. Unfortunately, I can't run that code because my
>>> project is in rails.
>>> But I didnot see the difference between you code and mine. What does the
>>> xml log file look like when you turn on "xml_curl debug_on"? Could you cut
>>> and paste them into  freeswitch pastebin?
>>>
>>> My code is cloned from
>>> http://wiki.freeswitch.org/wiki/Mod_xml_curl_Ruby_directory_example. I
>>> am not sure that the xml output may cause some problem.
>>>
>>> When I type "find_use_xml  id 1018 192.168.1.31", the FS ignored the
>>> response from web server. Instead, it retrieve the info from the local
>>> directory.
>>>
>>> freeswitch at proxy1.voiceserver.com> find_user_xml id 1018  192.168.1.31
>>> 2010-05-24 15:12:12.239449 [CONSOLE] mod_xml_curl.c:299 XML response is
>>> in /usr/local/freeswitch/
>>> scripts803ca405-c463-4deb-8121-982e1251831a.tmp.xml
>>> API CALL [find_user_xml(id 1018  192.168.1.31)] output:
>>> <user id="1018">
>>>       <params>
>>>           <param name="password" value="1234"></param>   <<<<===== the
>>> server response with "1018", "1234" is the local default setting
>>>
>>>           <param name="vm-password" value="1018"></param>
>>>
>>>     </params>
>>>
>>>       <variables>
>>>           <variable name="toll_allow"
>>> value="domestic,international,local"></variable>
>>>
>>>           <variable name="accountcode" value="1018"></variable>
>>>
>>>
>>>           <variable name="user_context" value="default"></variable>
>>>
>>>           <variable name="effective_caller_id_name" value="Extension
>>> 1018"></variable>
>>>
>>>           <variable name="effective_caller_id_number"
>>> value="1018"></variable>
>>>
>>>           <variable name="outbound_caller_id_name"
>>> value="MyPBX"></variable>
>>>
>>>           <variable name="outbound_caller_id_number"
>>> value="0000000000"></variable>
>>>
>>>           <variable name="callgroup" value="techsupport"></variable>
>>>
>>>     </variables>
>>>
>>>   </user>
>>>
>>> Below is the rails code to output the xml response.
>>> ==============================================
>>> ========
>>>
>>>   def directory
>>>
>>>     if params['user'].nil? || params['user'].empty?
>>>       ##if not looking for a user info, send 404 Not Found
>>>       @out_xml = Output404NotFound()
>>>     else
>>>       @username = params['user'].to_s
>>>       puts "username = " + @username
>>>       @domain = params['domain'].to_s
>>>       puts params
>>>       @dir=Directory.find_by_effective_caller_id_number(@username)
>>>       puts @dir
>>>       if @dir.nil?
>>>         @out_xml = Output404NotFound()
>>>       else
>>>         @out_xml = OutputXMLResponse()
>>>       end
>>>     end
>>>     puts "reply for directory query: " + @out_xml + "\n"
>>>   end
>>>
>>> private
>>>
>>>   def Output404NotFound
>>>     xml_text = "<?xml version=\"1.0\" encoding=\"UTF-8\"
>>> standalone=\"no\"?>
>>>                  <document type=\"freeswitch/xml\">
>>>                      <section name=\"result\">
>>>                           <result status=\"not found\" />
>>>                      </section>
>>>                  </document> "
>>>     return xml_text
>>>   end
>>>
>>>   def OutputXMLResponse
>>>     xml_text =
>>>         "<?xml version=\"1.0\" encoding=\"UTF-8\" standalone=\"no\"?>
>>>            <document type=\"freeswitch/xml\">
>>>              <section name=\"directory\" description=\"User Directory\">
>>>
>>>                  <domain name=\"#{@domain}\">
>>>                       <params>
>>>                          <param name=\"dial-string\"
>>> value=\"{presence_id=${dialed_user}@
>>> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}\"/>
>>>                       </params>
>>>
>>>                        <groups>
>>>                            <group name=\"default\">
>>>                                <users>
>>>                                    <user id=\"#{@username}\">
>>>                                      <params>
>>>                                         <param name=\"password\"
>>> value=\"" + @dir.password + "\"/>
>>>                                      </params>
>>>
>>>                                   </user>
>>>                                </users>
>>>                            </group>
>>>                          </groups>
>>>                    </domain>
>>>              </section>
>>>          </document> "
>>>      puts "reply for directory query: " + xml_text + "\n"
>>>      return xml_text
>>>   end
>>>
>>>
>>>
>>>
>>> On Mon, May 24, 2010 at 8:38 AM, Vitalii Colosov <vetali100 at gmail.com>wrote:
>>>
>>>> Just to avoid any misunderstanding - this is to check all other things
>>>> around freeswitch configuration.
>>>>
>>>> If this will work, it will be very easy to add database lookup for the
>>>> user trying to register:
>>>>
>>>>
>>>> $user = $_POST['user'];
>>>> $query="select password from your_users_table where user ='$user'";
>>>> $result=mysqli_query($db_connection, $query);
>>>>
>>>> ...get user and password from the result
>>>>
>>>> And then use $user and $password variables instead of hard-coded "1018",
>>>> "1018" in the code I provided.
>>>>
>>>> Really hope this helps, it works for me.
>>>>
>>>> Regards,
>>>> Vitalie
>>>>
>>>>
>>>> 2010/5/24 Vitalii Colosov <vetali100 at gmail.com>
>>>>
>>>>> Sorry there was an else related to the missing entry, i removed it.
>>>>>
>>>>> Please try another one.
>>>>>
>>>>> 2010/5/24 Vitalii Colosov <vetali100 at gmail.com>
>>>>>
>>>>> Please try to use my attached php script, if you can.
>>>>>>
>>>>>> I changed it to fit your data (user 1018, pass: 1018), but I did not
>>>>>> check for any syntax error (hope it does not have :-) ).
>>>>>>
>>>>>> Please reply back on result.
>>>>>>
>>>>>> Regards,
>>>>>>
>>>>>> Vitalie
>>>>>>
>>>>>>
>>>>>> 2010/5/24 Erwin Davis <davis.erwin at gmail.com>
>>>>>>
>>>>>>> yeah, my code replied with line 29-46. I donot know where the rest of
>>>>>>> html code was generated.
>>>>>>>
>>>>>>> On Mon, May 24, 2010 at 10:34 AM, Milena <testeador01 at gmail.com>wrote:
>>>>>>>
>>>>>>>> Hello,
>>>>>>>> You're not meant to send a whole website full of html tags as a
>>>>>>>> response.
>>>>>>>> Just reply the part of the lines 29 - 46 on your pastebin and get
>>>>>>>> rid of the rest.
>>>>>>>>
>>>>>>>>
>>>>>>>> 2010,
>>>>>>>> /5/24 Erwin Davis <davis.erwin at gmail.com>
>>>>>>>>
>>>>>>>> Hi, I am stuck, My console log is in
>>>>>>>>> http://pastebin.freeswitch.org/13030 and my xml log is in
>>>>>>>>> http://pastebin.freeswitch.org/13029.  I guess that the problem
>>>>>>>>> should be pretty small but I just can not figure it out. -:(  Thanks for
>>>>>>>>> your help.
>>>>>>>>>
>>>>>>>>> On Mon, May 24, 2010 at 8:12 AM, Vitalii Colosov <
>>>>>>>>> vetali100 at gmail.com> wrote:
>>>>>>>>>
>>>>>>>>>> Hi Erwin,
>>>>>>>>>> Maybe you need to add the first line <?xml version="1.0"
>>>>>>>>>> encoding="UTF-8" standalone="no"?>
>>>>>>>>>>
>>>>>>>>>> This is what I have and it works perfectly, tested right now:
>>>>>>>>>>
>>>>>>>>>> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>>>>>>>>>> <document type="freeswitch/xml">
>>>>>>>>>>   <section name="directory">
>>>>>>>>>>     <domain name="$${domain}">
>>>>>>>>>>       <params>
>>>>>>>>>>         <param name="dial-string"
>>>>>>>>>> value="{presence_id=${dialed_user}@
>>>>>>>>>> ${dialed_domain}}${sofia_contact(${dialed_user}@
>>>>>>>>>> ${dialed_domain})}"/>
>>>>>>>>>>       </params>
>>>>>>>>>>       <groups>
>>>>>>>>>>         <group name="default">
>>>>>>>>>>           <users>
>>>>>>>>>>             <user id="1000">
>>>>>>>>>>               <params>
>>>>>>>>>>                 <param name="a1-hash" value="somehashvalue"/>
>>>>>>>>>>                 <param name="vm-a1-hash" value="somehashvalue"/>
>>>>>>>>>>               </params>
>>>>>>>>>>               <variables>
>>>>>>>>>>                 <variable name="user_context" value="default"/>
>>>>>>>>>>                 <variable name="accountcode" value="1000"/>
>>>>>>>>>>               </variables>
>>>>>>>>>>             </user>
>>>>>>>>>>           </users>
>>>>>>>>>>         </group>
>>>>>>>>>>       </groups>
>>>>>>>>>>     </domain>
>>>>>>>>>>   </section>
>>>>>>>>>> </document>
>>>>>>>>>>
>>>>>>>>>> Regards,
>>>>>>>>>> Vitalie
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
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
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From msc at freeswitch.org  Mon May 24 17:07:41 2010
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 24 May 2010 17:07:41 -0700
Subject: [Freeswitch-users] Question about inbound ACL
In-Reply-To: <AANLkTimAepVJa1kK9Qfut7jRZK1q4zjIjzUfsvu18nZt@mail.gmail.com>
References: <AANLkTimAepVJa1kK9Qfut7jRZK1q4zjIjzUfsvu18nZt@mail.gmail.com>
Message-ID: <AANLkTilSVUR5I2Pd9xwbneXnWSB5VpcuOAv8bGwV34Qo@mail.gmail.com>

On Mon, May 24, 2010 at 1:45 PM, RR <ranjtech at gmail.com> wrote:

> Hello List,
>
> I have tried to read as much as I could and try out a bunch of things but
> it doesn't seem to be working. The scenario is that we have a FS setup as a
> call distributor to our internal SIP servers. The calls come into the FS
> when people call any of the 6000 or so DIDs we own. These DIDs are through
> various providers who's IP addresses we know (obviously). I want to receive
> / process only calls from these external IP addresses and nothing else. So I
> did the following in acl.conf.xml file
>
>         <list name="DIDProviders" default="deny">
>           <node type="allow" cidr="64.XXX.XXX.XXX/32"/>
>           <node type="allow" cidr="67.XXX.XXX.XXX/30"/>
>            etc...
>         </list>
>
> then I went into the $FSHOME/conf/sip_profiles/internal.xml and did the
> following:
>
> <param name="apply-inbound-acl" value="DIDProviders"/>
>
> but I still keep seeing calls from other IPs that are not in the
> "DIDProviders" list getting through.
>
> What else do I need to do to prevent this? Should this not be in
> internal.xml but in external.xml?
>

I have to ask: did you do this at the fs_cli:
reloadacl reloadxml

Need to make sure that your ACLs got reloaded. If so, then the next step is
to turn on full console debugging (which is on by default when using fs_cli)
and try a call. Capture the complete log for a call and drop it in pastebin.
Redact any private information and then paste the link in this thread. We'll
take a look.

-MC


>
> TIA
> \RR
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Mon May 24 17:11:03 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 24 May 2010 20:11:03 -0400
Subject: [Freeswitch-users] Mod_skinny: Call for testing!
In-Reply-To: <4BFA5A12.901@puzzled.xs4all.nl>
References: <AANLkTik8hHRtOci8qTXmZVHOWvVxRg-OPm5zyRAMrWEe@mail.gmail.com>
	<4BFA5A12.901@puzzled.xs4all.nl>
Message-ID: <F804849B-822F-4AC5-998E-577565BFE516@jerris.com>

Mathieu, we have a define for the printf specifier for switch_size_t, because it is different on different os's:

switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_NOTICE, "Session %" SWITCH_SIZE_T_FMT " (%s) Ended\n",

you can use it like that, or optionally, if appropriate, cast the value to the right type when passing it to printf style commands.

Mike

On May 24, 2010, at 6:50 AM, Patrick wrote:

> On 05/21/2010 02:54 AM, Mathieu Parent wrote:
> [snip]
> 
> Hi Mathieu,
> 
> On Fedora 12 and FreeSWITCH git rev f89cbdd... (from yesterday) I get 
> the following error:
> 
> quiet_libtool: compile:  gcc -DHAVE_CONFIG_H -I. 
> -I../../../../src/include -I../../../../libs/xmlrpc-c 
> -I/home/patrick/redhat/BUILD/freeswitch/src/include 
> -I/home/patrick/redhat/BUILD/freeswitch/src/include 
> -I/home/patrick/redhat/BUILD/freeswitch/libs/libteletone/src -fPIC 
> -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 
> -I/home/patrick/redhat/BUILD/freeswitch/src/include 
> -I/home/patrick/redhat/BUILD/freeswitch/src/include 
> -I/home/patrick/redhat/BUILD/freeswitch/libs/libteletone/src -fPIC 
> -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 
> -DHAVE_VISIBILITY=1 -g -ggdb -Wall -std=c99 -pedantic 
> -Wdeclaration-after-statement -g -ggdb -MT 
> mod_skinny_la-skinny_server.lo -MD -MP -MF 
> .deps/mod_skinny_la-skinny_server.Tpo -c skinny_server.c  -fPIC -DPIC -o 
> .libs/mod_skinny_la-skinny_server.o
> cc1: warnings being treated as errors
> skinny_server.c: In function ?skinny_handle_register?:
> skinny_server.c:933: error: format ?%d? expects type ?int?, but argument 
> 11 has type ?long unsigned int?
> skinny_server.c: In function ?skinny_handle_port_message?:
> skinny_server.c:1084: error: format ?%d? expects type ?int?, but 
> argument 11 has type ?long unsigned int?
> skinny_server.c: In function ?skinny_handle_keypad_button_message?:
> skinny_server.c:1104: error: format ?%d? expects type ?int?, but 
> argument 11 has type ?long unsigned int?
> skinny_server.c: In function ?skinny_handle_stimulus_message?:
> skinny_server.c:1164: error: format ?%d? expects type ?int?, but 
> argument 11 has type ?long unsigned int?
> skinny_server.c: In function ?skinny_handle_speed_dial_stat_request?:
> skinny_server.c:1271: error: format ?%d? expects type ?int?, but 
> argument 11 has type ?long unsigned int?
> skinny_server.c: In function ?skinny_handle_line_stat_request?:
> skinny_server.c:1291: error: format ?%d? expects type ?int?, but 
> argument 11 has type ?long unsigned int?
> skinny_server.c: In function ?skinny_handle_capabilities_response?:
> skinny_server.c:1501: error: format ?%d? expects type ?int?, but 
> argument 11 has type ?long unsigned int?
> skinny_server.c:1509: error: format ?%d? expects type ?int?, but 
> argument 11 has type ?long unsigned int?
> skinny_server.c: In function ?skinny_handle_alarm?:
> skinny_server.c:1548: error: format ?%d? expects type ?int?, but 
> argument 11 has type ?long unsigned int?
> skinny_server.c: In function 
> ?skinny_handle_open_receive_channel_ack_message?:
> skinny_server.c:1571: error: format ?%d? expects type ?int?, but 
> argument 11 has type ?long unsigned int?
> skinny_server.c: In function ?skinny_handle_soft_key_event_message?:
> skinny_server.c:1722: error: format ?%d? expects type ?int?, but 
> argument 11 has type ?long unsigned int?
> skinny_server.c: In function ?skinny_headset_status_message?:
> skinny_server.c:1847: error: format ?%d? expects type ?int?, but 
> argument 11 has type ?long unsigned int?
> skinny_server.c: In function 
> ?skinny_handle_register_available_lines_message?:
> skinny_server.c:1855: error: format ?%d? expects type ?int?, but 
> argument 11 has type ?long unsigned int?
> skinny_server.c: In function ?skinny_handle_service_url_stat_request?:
> skinny_server.c:1866: error: format ?%d? expects type ?int?, but 
> argument 11 has type ?long unsigned int?
> skinny_server.c: In function ?skinny_handle_feature_stat_request?:
> skinny_server.c:1886: error: format ?%d? expects type ?int?, but 
> argument 11 has type ?long unsigned int?
> make: *** [mod_skinny_la-skinny_server.lo] Error 1



From wangdq.no1 at gmail.com  Mon May 24 17:23:00 2010
From: wangdq.no1 at gmail.com (daqiang wang)
Date: Tue, 25 May 2010 08:23:00 +0800
Subject: [Freeswitch-users] how can i get the variable
	"Caller-Channel-Answered-Time" in the API command ?
Message-ID: <AANLkTin93HnNuX1dCJAc21MlVaZJRQXtTTDt9Z1VgA1K@mail.gmail.com>


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From brian at freeswitch.org  Mon May 24 17:37:19 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 24 May 2010 19:37:19 -0500
Subject: [Freeswitch-users] how can i get the variable
	"Caller-Channel-Answered-Time" in the API command ?
In-Reply-To: <AANLkTin93HnNuX1dCJAc21MlVaZJRQXtTTDt9Z1VgA1K@mail.gmail.com>
References: <AANLkTin93HnNuX1dCJAc21MlVaZJRQXtTTDt9Z1VgA1K@mail.gmail.com>
Message-ID: <0AAC52CD-5766-46D1-B1C9-058E09172148@freeswitch.org>

How about you learn to use the EMAIL body instead of the subject to type your question out.  

What exactly are you trying to accomplish?  Are you trying to do billing inline in a script with the caller?

/b

On May 24, 2010, at 7:23 PM, daqiang wang wrote:

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



From msc at freeswitch.org  Mon May 24 17:39:57 2010
From: msc at freeswitch.org (Michael S Collins)
Date: Mon, 24 May 2010 17:39:57 -0700
Subject: [Freeswitch-users] how can i get the variable
	"Caller-Channel-Answered-Time" in the API command ?
In-Reply-To: <AANLkTin93HnNuX1dCJAc21MlVaZJRQXtTTDt9Z1VgA1K@mail.gmail.com>
References: <AANLkTin93HnNuX1dCJAc21MlVaZJRQXtTTDt9Z1VgA1K@mail.gmail.com>
Message-ID: <A3F6EE8A-3700-4F4D-889D-0F9A037F061D@freeswitch.org>

API uuid_getvar

Sent from my iPhone

On May 24, 2010, at 5:23 PM, daqiang wang <wangdq.no1 at gmail.com> wrote:

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


From wangdq.no1 at gmail.com  Mon May 24 18:35:23 2010
From: wangdq.no1 at gmail.com (daqiang wang)
Date: Tue, 25 May 2010 09:35:23 +0800
Subject: [Freeswitch-users] how can i get the variable
	"Caller-Channel-Answered-Time" in the API command ?
In-Reply-To: <A3F6EE8A-3700-4F4D-889D-0F9A037F061D@freeswitch.org>
References: <AANLkTin93HnNuX1dCJAc21MlVaZJRQXtTTDt9Z1VgA1K@mail.gmail.com>
	<A3F6EE8A-3700-4F4D-889D-0F9A037F061D@freeswitch.org>
Message-ID: <AANLkTimB9prvFKF6jy7sK5CKyHOFaZ3m_3Ao2ewzAdGx@mail.gmail.com>

I use uuid_getvar can't get the variable value.
but I can get "sip_user_name".  why ?


2010/5/25 Michael S Collins <msc at freeswitch.org>

> API uuid_getvar
>
> Sent from my iPhone
>
> On May 24, 2010, at 5:23 PM, daqiang wang <wangdq.no1 at gmail.com> wrote:
>
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Mon May 24 18:43:29 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 24 May 2010 20:43:29 -0500
Subject: [Freeswitch-users] how can i get the variable
	"Caller-Channel-Answered-Time" in the API command ?
In-Reply-To: <AANLkTimB9prvFKF6jy7sK5CKyHOFaZ3m_3Ao2ewzAdGx@mail.gmail.com>
References: <AANLkTin93HnNuX1dCJAc21MlVaZJRQXtTTDt9Z1VgA1K@mail.gmail.com>
	<A3F6EE8A-3700-4F4D-889D-0F9A037F061D@freeswitch.org>
	<AANLkTimB9prvFKF6jy7sK5CKyHOFaZ3m_3Ao2ewzAdGx@mail.gmail.com>
Message-ID: <E518821E-43C8-4DBC-95B4-78A8EEC2B555@freeswitch.org>

What are you doing exactly?  And how are you doing it?  Some of those variables related to call times aren't there till the call is hung up.

/b

On May 24, 2010, at 8:35 PM, daqiang wang wrote:

> I use uuid_getvar can't get the variable value. 
> but I can get "sip_user_name".  why ? 
> 
> 
> 2010/5/25 Michael S Collins <msc at freeswitch.org>
> API uuid_getvar
> 
> Sent from my iPhone
> 
> On May 24, 2010, at 5:23 PM, daqiang wang <wangdq.no1 at gmail.com> wrote:

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From brian at freeswitch.org  Mon May 24 18:44:42 2010
From: brian at freeswitch.org (Brian West)
Date: Mon, 24 May 2010 20:44:42 -0500
Subject: [Freeswitch-users] how can i get the variable
	"Caller-Channel-Answered-Time" in the API command ?
In-Reply-To: <AANLkTimB9prvFKF6jy7sK5CKyHOFaZ3m_3Ao2ewzAdGx@mail.gmail.com>
References: <AANLkTin93HnNuX1dCJAc21MlVaZJRQXtTTDt9Z1VgA1K@mail.gmail.com>
	<A3F6EE8A-3700-4F4D-889D-0F9A037F061D@freeswitch.org>
	<AANLkTimB9prvFKF6jy7sK5CKyHOFaZ3m_3Ao2ewzAdGx@mail.gmail.com>
Message-ID: <4F48A9D3-EA8E-4FAF-B6C5-83FC45C3C03A@freeswitch.org>

Open up src/switch_caller.c

freeswitch at internal> uuid_getvar d63aa129-0430-4ed6-9b06-2d66888830e8 answered_time
1274738592435838

/b

On May 24, 2010, at 8:35 PM, daqiang wang wrote:

> I use uuid_getvar can't get the variable value. 
> but I can get "sip_user_name".  why ? 
> 
> 
> 2010/5/25 Michael S Collins <msc at freeswitch.org>
> API uuid_getvar
> 
> Sent from my iPhone
> 
> On May 24, 2010, at 5:23 PM, daqiang wang <wangdq.no1 at gmail.com> wrote:

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From wangdq.no1 at gmail.com  Mon May 24 19:07:27 2010
From: wangdq.no1 at gmail.com (daqiang wang)
Date: Tue, 25 May 2010 10:07:27 +0800
Subject: [Freeswitch-users] how can i get the variable
	"Caller-Channel-Answered-Time" in the API command ?
In-Reply-To: <4F48A9D3-EA8E-4FAF-B6C5-83FC45C3C03A@freeswitch.org>
References: <AANLkTin93HnNuX1dCJAc21MlVaZJRQXtTTDt9Z1VgA1K@mail.gmail.com>
	<A3F6EE8A-3700-4F4D-889D-0F9A037F061D@freeswitch.org>
	<AANLkTimB9prvFKF6jy7sK5CKyHOFaZ3m_3Ao2ewzAdGx@mail.gmail.com>
	<4F48A9D3-EA8E-4FAF-B6C5-83FC45C3C03A@freeswitch.org>
Message-ID: <AANLkTimiGZNLyy-d6HatiZYd7DiWc1nNhRAO5MH9uXsz@mail.gmail.com>

thank you very much !
I get it .

2010/5/25 Brian West <brian at freeswitch.org>

> Open up src/switch_caller.c
>
> freeswitch at internal> uuid_getvar d63aa129-0430-4ed6-9b06-2d66888830e8
> answered_time
> 1274738592435838
>
> /b
>
> On May 24, 2010, at 8:35 PM, daqiang wang wrote:
>
> I use uuid_getvar can't get the variable value.
> but I can get "sip_user_name".  why ?
>
>
> 2010/5/25 Michael S Collins <msc at freeswitch.org>
>
>> API uuid_getvar
>>
>> Sent from my iPhone
>>
>> On May 24, 2010, at 5:23 PM, daqiang wang <wangdq.no1 at gmail.com> wrote:
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jeff at jefflenk.com  Mon May 24 21:11:46 2010
From: jeff at jefflenk.com (Jeff Lenk)
Date: Mon, 24 May 2010 21:11:46 -0700 (PDT)
Subject: [Freeswitch-users] SVN Mirror - Closed
In-Reply-To: <7B8D2674FE3349708310970BEA69EF97@bp1.ad.bp.com>
References: <B3171C43FAC94239822FAB3CD8DC2D7E@bp1.ad.bp.com>
	<20100522100933.GA8450@jdc.jasonjgw.net>
	<D34CD729D65140D0B1ED3C9726488BE6@bp1.ad.bp.com>
	<20100522111309.GA8783@jdc.jasonjgw.net>
	<5F9E6D41F9C64AA599D5530C9F21DE00@bp1.ad.bp.com>
	<326A92F1557241BA8C21B8FA73FD217B@dell9400>
	<D701F9999818488781541CCCBE0EDC60@bp1.ad.bp.com>
	<7B8D2674FE3349708310970BEA69EF97@bp1.ad.bp.com>
Message-ID: <1274760706129-5096801.post@n2.nabble.com>


The current version of msysgit and tortoisegit work real well.

Install mysgit with autocrlf=false and everything else default.

Then follow these instructions-
http://wiki.freeswitch.org/wiki/Download_FreeSWITCH#Simple_Clone_-_Windows
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/SVN-Mirror-tp5087575p5096801.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From nagalenoj at gmail.com  Mon May 24 23:38:18 2010
From: nagalenoj at gmail.com (Nagalenoj H.)
Date: Tue, 25 May 2010 12:08:18 +0530
Subject: [Freeswitch-users] IRC log for #openzap
Message-ID: <AANLkTimnxUUYtYdrk7_3T9_GNFn0BusFxkXO4cdZu5yb@mail.gmail.com>

Hi friends,
     Where is the openzap channel's IRC log available? Has someone tried
downloading it already?

-- 
Regards,
Nagalenoj H.
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From math.parent at gmail.com  Tue May 25 00:42:35 2010
From: math.parent at gmail.com (Mathieu Parent)
Date: Tue, 25 May 2010 09:42:35 +0200
Subject: [Freeswitch-users] Mod_skinny: Call for testing!
In-Reply-To: <F804849B-822F-4AC5-998E-577565BFE516@jerris.com>
References: <AANLkTik8hHRtOci8qTXmZVHOWvVxRg-OPm5zyRAMrWEe@mail.gmail.com>
	<4BFA5A12.901@puzzled.xs4all.nl>
	<F804849B-822F-4AC5-998E-577565BFE516@jerris.com>
Message-ID: <AANLkTinwptx6T23LN0IIM9nCVl19g-M7aYl2o8i7-eI1@mail.gmail.com>

On Tue, May 25, 2010 at 2:11 AM, Michael Jerris <mike at jerris.com> wrote:
> Mathieu, we have a define for the printf specifier for switch_size_t, because it is different on different os's:
>
> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_NOTICE, "Session %" SWITCH_SIZE_T_FMT " (%s) Ended\n",
>
> you can use it like that, or optionally, if appropriate, cast the value to the right type when passing it to printf style commands.
>

Thanks Mike. I thought I had already done this but can find where it went.

I also need a function for big-endian platforms to convert native
integer to little endian and the reverse. Those are simple to
implement and I can do this, but I don't know where to put them
(switch_utils.{h,c}?) and how to name them (switch_htol32,
switch_ltoh32? or switch_native_to_little_endian,
switch_to_little?...) ; or maybe they already exists?

Patrick, his is solved on a2ceff1. Can you test again? For reference,
this is MODSKINNY-5 on Jira.

Mathieu


From frank at impactfax.com  Tue May 25 03:13:18 2010
From: frank at impactfax.com (Frank @ Impact)
Date: Tue, 25 May 2010 06:13:18 -0400
Subject: [Freeswitch-users] sip_cid_type settings
Message-ID: <01991B335AA9418CB32018A477F5857F@ws4>

I can only find documentation on setting this value at the bridge level.
 
Is there a way to control this setting in the gateway profile instead of
having to do it in the bridge setup? 
 
 
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From david.ponzone at gmail.com  Tue May 25 03:43:05 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Tue, 25 May 2010 12:43:05 +0200
Subject: [Freeswitch-users] sip_cid_type settings
In-Reply-To: <01991B335AA9418CB32018A477F5857F@ws4>
References: <01991B335AA9418CB32018A477F5857F@ws4>
Message-ID: <AABFAAFF-0CC1-46C6-A03E-E338906FB700@gmail.com>

Frank,

as far as I know, there is not.
You can probably simulate that by using a custom variable in the  
gateway profile, like accountcode, and then matching that at the  
beginning of your dialplan to set the sip_cid_type accordingly.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 25/05/2010 ? 12:13, Frank @ Impact a ?crit :

> I can only find documentation on setting this value at the bridge  
> level.
>
> Is there a way to control this setting in the gateway profile  
> instead of having to do it in the bridge setup?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From ben at langfeld.co.uk  Tue May 25 05:52:03 2010
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Tue, 25 May 2010 13:52:03 +0100
Subject: [Freeswitch-users] curl_xml
In-Reply-To: <AANLkTimHilHi_YnSQj58IjUHCPxcwxf4wcNfhXlOSmuI@mail.gmail.com>
References: <AANLkTiltmx-qzAAgkQjUcCAGsAMp9_X573zbQic24pf7@mail.gmail.com>
	<AANLkTim6sWASDmj8w6KF9gsgB8joPKZC6VX34HpEOCdE@mail.gmail.com>
	<AANLkTimWDqSf2a7ZacI6q_4AQP5wm1sJImEKZBX3QzbC@mail.gmail.com>
	<AANLkTilAIOnMu1bmJBOTsCvrnSfytRE8tenp6JEVTMc-@mail.gmail.com>
	<AANLkTin9OKJNURCKZ7__EMGyo1_4WYXufdQhDmYAcJOb@mail.gmail.com>
	<AANLkTimR6zWJX3bwGCuvt5efDRlj3qY_Xiisn3doGupD@mail.gmail.com>
	<AANLkTinsj5G5R2fCYFmDyhEmIu4empwzE0psrUU-q6fL@mail.gmail.com>
	<AANLkTilU7uEEjVhVVo6OAObi7Bq9o2cAmC1rmhrlglr3@mail.gmail.com>
	<AANLkTildFYq1rtaLicpS7kFnrRWk5W1GxQNMyFJMy9tL@mail.gmail.com>
	<AANLkTimHilHi_YnSQj58IjUHCPxcwxf4wcNfhXlOSmuI@mail.gmail.com>
Message-ID: <AANLkTim8gkyKDOzqvwLLXvkoJ_YLfcD69GjBBt7SARa4@mail.gmail.com>

You need to specify a new rails layout that is empty. You are currently
loading the application layout and rendering your XML in the body.

Regards,
Ben Langfeld


On Tue, May 25, 2010 at 12:36 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> I am jumping in here because I am concerned that my browser will crash if
> this thread gets any bigger:
>
> Perhaps if you change your point of view you will find it faster.
> You have multiple accounts from other users that it works, but i get the
> feeling you believe in the back of your mind that it's broken somehow.  If
> you change your mindset to "I know I am doing something wrong and all I need
> to do is figure out what it is", probably you will be able to spot the
> mistake easier.
>
> Thank you everyone who has helped here.
>
>
>
>
> On Mon, May 24, 2010 at 3:47 PM, Erwin Davis <davis.erwin at gmail.com>wrote:
>
>> Hello,
>>  Can anyone post a xml log file when you turn on "xml_curl debug_on"? I
>> hope to compare with the my log file with a valid one? Thanks,
>>
>> Regards,
>>
>> e
>>
>>
>> On Mon, May 24, 2010 at 3:45 PM, Vitalii Colosov <vetali100 at gmail.com>wrote:
>>
>>> Sorry, I never used rubby, no help from my side here...
>>>
>>> This is my XML response, it's not so long to put it in pastebin, also I
>>> never used pastebin yet:
>>>
>>> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>>> <document type="freeswitch/xml">
>>>   <section name="directory">
>>>     <domain name="$${domain}">
>>>       <params>
>>>         <param name="dial-string" value="{presence_id=${dialed_user}@
>>> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>>>       </params>
>>>       <groups>
>>>         <group name="default">
>>>           <users>
>>>             <user id="1000">
>>>               <params>
>>>                 <param name="a1-hash" value="somehashvalue"/>
>>>                 <param name="vm-a1-hash" value="somehashvalue"/>
>>>               </params>
>>>               <variables>
>>>                 <variable name="user_context" value="default"/>
>>>                 <variable name="accountcode" value="1000"/>
>>>               </variables>
>>>             </user>
>>>           </users>
>>>         </group>
>>>       </groups>
>>>     </domain>
>>>   </section>
>>> </document>
>>>
>>> Regards,
>>> Vitalie
>>>
>>> 2010/5/24 Erwin Davis <davis.erwin at gmail.com>
>>>
>>>> Hi, Vitalli,
>>>>
>>>> Thanks for your PHP code. Unfortunately, I can't run that code because
>>>> my project is in rails.
>>>> But I didnot see the difference between you code and mine. What does the
>>>> xml log file look like when you turn on "xml_curl debug_on"? Could you cut
>>>> and paste them into  freeswitch pastebin?
>>>>
>>>> My code is cloned from
>>>> http://wiki.freeswitch.org/wiki/Mod_xml_curl_Ruby_directory_example. I
>>>> am not sure that the xml output may cause some problem.
>>>>
>>>> When I type "find_use_xml  id 1018 192.168.1.31", the FS ignored the
>>>> response from web server. Instead, it retrieve the info from the local
>>>> directory.
>>>>
>>>> freeswitch at proxy1.voiceserver.com> find_user_xml id 1018  192.168.1.31
>>>> 2010-05-24 15:12:12.239449 [CONSOLE] mod_xml_curl.c:299 XML response is
>>>> in /usr/local/freeswitch/
>>>> scripts803ca405-c463-4deb-8121-982e1251831a.tmp.xml
>>>> API CALL [find_user_xml(id 1018  192.168.1.31)] output:
>>>> <user id="1018">
>>>>       <params>
>>>>           <param name="password" value="1234"></param>   <<<<===== the
>>>> server response with "1018", "1234" is the local default setting
>>>>
>>>>           <param name="vm-password" value="1018"></param>
>>>>
>>>>     </params>
>>>>
>>>>       <variables>
>>>>           <variable name="toll_allow"
>>>> value="domestic,international,local"></variable>
>>>>
>>>>           <variable name="accountcode" value="1018"></variable>
>>>>
>>>>
>>>>           <variable name="user_context" value="default"></variable>
>>>>
>>>>           <variable name="effective_caller_id_name" value="Extension
>>>> 1018"></variable>
>>>>
>>>>           <variable name="effective_caller_id_number"
>>>> value="1018"></variable>
>>>>
>>>>           <variable name="outbound_caller_id_name"
>>>> value="MyPBX"></variable>
>>>>
>>>>           <variable name="outbound_caller_id_number"
>>>> value="0000000000"></variable>
>>>>
>>>>           <variable name="callgroup" value="techsupport"></variable>
>>>>
>>>>     </variables>
>>>>
>>>>   </user>
>>>>
>>>> Below is the rails code to output the xml response.
>>>> ==============================================
>>>> ========
>>>>
>>>>   def directory
>>>>
>>>>     if params['user'].nil? || params['user'].empty?
>>>>       ##if not looking for a user info, send 404 Not Found
>>>>       @out_xml = Output404NotFound()
>>>>     else
>>>>       @username = params['user'].to_s
>>>>       puts "username = " + @username
>>>>       @domain = params['domain'].to_s
>>>>       puts params
>>>>       @dir=Directory.find_by_effective_caller_id_number(@username)
>>>>       puts @dir
>>>>       if @dir.nil?
>>>>         @out_xml = Output404NotFound()
>>>>       else
>>>>         @out_xml = OutputXMLResponse()
>>>>       end
>>>>     end
>>>>     puts "reply for directory query: " + @out_xml + "\n"
>>>>   end
>>>>
>>>> private
>>>>
>>>>   def Output404NotFound
>>>>     xml_text = "<?xml version=\"1.0\" encoding=\"UTF-8\"
>>>> standalone=\"no\"?>
>>>>                  <document type=\"freeswitch/xml\">
>>>>                      <section name=\"result\">
>>>>                           <result status=\"not found\" />
>>>>                      </section>
>>>>                  </document> "
>>>>     return xml_text
>>>>   end
>>>>
>>>>   def OutputXMLResponse
>>>>     xml_text =
>>>>         "<?xml version=\"1.0\" encoding=\"UTF-8\" standalone=\"no\"?>
>>>>            <document type=\"freeswitch/xml\">
>>>>              <section name=\"directory\" description=\"User Directory\">
>>>>
>>>>                  <domain name=\"#{@domain}\">
>>>>                       <params>
>>>>                          <param name=\"dial-string\"
>>>> value=\"{presence_id=${dialed_user}@
>>>> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}\"/>
>>>>                       </params>
>>>>
>>>>                        <groups>
>>>>                            <group name=\"default\">
>>>>                                <users>
>>>>                                    <user id=\"#{@username}\">
>>>>                                      <params>
>>>>                                         <param name=\"password\"
>>>> value=\"" + @dir.password + "\"/>
>>>>                                      </params>
>>>>
>>>>                                   </user>
>>>>                                </users>
>>>>                            </group>
>>>>                          </groups>
>>>>                    </domain>
>>>>              </section>
>>>>          </document> "
>>>>      puts "reply for directory query: " + xml_text + "\n"
>>>>      return xml_text
>>>>   end
>>>>
>>>>
>>>>
>>>>
>>>> On Mon, May 24, 2010 at 8:38 AM, Vitalii Colosov <vetali100 at gmail.com>wrote:
>>>>
>>>>> Just to avoid any misunderstanding - this is to check all other things
>>>>> around freeswitch configuration.
>>>>>
>>>>> If this will work, it will be very easy to add database lookup for the
>>>>> user trying to register:
>>>>>
>>>>>
>>>>> $user = $_POST['user'];
>>>>> $query="select password from your_users_table where user ='$user'";
>>>>> $result=mysqli_query($db_connection, $query);
>>>>>
>>>>> ...get user and password from the result
>>>>>
>>>>> And then use $user and $password variables instead
>>>>> of hard-coded "1018", "1018" in the code I provided.
>>>>>
>>>>> Really hope this helps, it works for me.
>>>>>
>>>>> Regards,
>>>>> Vitalie
>>>>>
>>>>>
>>>>> 2010/5/24 Vitalii Colosov <vetali100 at gmail.com>
>>>>>
>>>>>> Sorry there was an else related to the missing entry, i removed it.
>>>>>>
>>>>>> Please try another one.
>>>>>>
>>>>>> 2010/5/24 Vitalii Colosov <vetali100 at gmail.com>
>>>>>>
>>>>>> Please try to use my attached php script, if you can.
>>>>>>>
>>>>>>> I changed it to fit your data (user 1018, pass: 1018), but I did not
>>>>>>> check for any syntax error (hope it does not have :-) ).
>>>>>>>
>>>>>>> Please reply back on result.
>>>>>>>
>>>>>>> Regards,
>>>>>>>
>>>>>>> Vitalie
>>>>>>>
>>>>>>>
>>>>>>> 2010/5/24 Erwin Davis <davis.erwin at gmail.com>
>>>>>>>
>>>>>>>> yeah, my code replied with line 29-46. I donot know where the rest
>>>>>>>> of html code was generated.
>>>>>>>>
>>>>>>>> On Mon, May 24, 2010 at 10:34 AM, Milena <testeador01 at gmail.com>wrote:
>>>>>>>>
>>>>>>>>> Hello,
>>>>>>>>> You're not meant to send a whole website full of html tags as a
>>>>>>>>> response.
>>>>>>>>> Just reply the part of the lines 29 - 46 on your pastebin and get
>>>>>>>>> rid of the rest.
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> 2010,
>>>>>>>>> /5/24 Erwin Davis <davis.erwin at gmail.com>
>>>>>>>>>
>>>>>>>>> Hi, I am stuck, My console log is in
>>>>>>>>>> http://pastebin.freeswitch.org/13030 and my xml log is in
>>>>>>>>>> http://pastebin.freeswitch.org/13029.  I guess that the problem
>>>>>>>>>> should be pretty small but I just can not figure it out. -:(  Thanks for
>>>>>>>>>> your help.
>>>>>>>>>>
>>>>>>>>>> On Mon, May 24, 2010 at 8:12 AM, Vitalii Colosov <
>>>>>>>>>> vetali100 at gmail.com> wrote:
>>>>>>>>>>
>>>>>>>>>>> Hi Erwin,
>>>>>>>>>>> Maybe you need to add the first line <?xml version="1.0"
>>>>>>>>>>> encoding="UTF-8" standalone="no"?>
>>>>>>>>>>>
>>>>>>>>>>> This is what I have and it works perfectly, tested right now:
>>>>>>>>>>>
>>>>>>>>>>> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
>>>>>>>>>>> <document type="freeswitch/xml">
>>>>>>>>>>>   <section name="directory">
>>>>>>>>>>>     <domain name="$${domain}">
>>>>>>>>>>>       <params>
>>>>>>>>>>>         <param name="dial-string"
>>>>>>>>>>> value="{presence_id=${dialed_user}@
>>>>>>>>>>> ${dialed_domain}}${sofia_contact(${dialed_user}@
>>>>>>>>>>> ${dialed_domain})}"/>
>>>>>>>>>>>       </params>
>>>>>>>>>>>       <groups>
>>>>>>>>>>>         <group name="default">
>>>>>>>>>>>           <users>
>>>>>>>>>>>             <user id="1000">
>>>>>>>>>>>               <params>
>>>>>>>>>>>                 <param name="a1-hash" value="somehashvalue"/>
>>>>>>>>>>>                 <param name="vm-a1-hash" value="somehashvalue"/>
>>>>>>>>>>>               </params>
>>>>>>>>>>>               <variables>
>>>>>>>>>>>                 <variable name="user_context" value="default"/>
>>>>>>>>>>>                 <variable name="accountcode" value="1000"/>
>>>>>>>>>>>               </variables>
>>>>>>>>>>>             </user>
>>>>>>>>>>>           </users>
>>>>>>>>>>>         </group>
>>>>>>>>>>>       </groups>
>>>>>>>>>>>     </domain>
>>>>>>>>>>>   </section>
>>>>>>>>>>> </document>
>>>>>>>>>>>
>>>>>>>>>>> Regards,
>>>>>>>>>>> Vitalie
>>>>>>>>>>>
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:
>>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:
>>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> 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
>
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>
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From mike at jerris.com  Tue May 25 05:57:05 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 25 May 2010 08:57:05 -0400
Subject: [Freeswitch-users] Mod_skinny: Call for testing!
In-Reply-To: <AANLkTinwptx6T23LN0IIM9nCVl19g-M7aYl2o8i7-eI1@mail.gmail.com>
References: <AANLkTik8hHRtOci8qTXmZVHOWvVxRg-OPm5zyRAMrWEe@mail.gmail.com>
	<4BFA5A12.901@puzzled.xs4all.nl>
	<F804849B-822F-4AC5-998E-577565BFE516@jerris.com>
	<AANLkTinwptx6T23LN0IIM9nCVl19g-M7aYl2o8i7-eI1@mail.gmail.com>
Message-ID: <5212488E-3EB9-43D9-85C9-99D809DA27D8@jerris.com>


On May 25, 2010, at 3:42 AM, Mathieu Parent wrote:

> On Tue, May 25, 2010 at 2:11 AM, Michael Jerris <mike at jerris.com> wrote:
>> Mathieu, we have a define for the printf specifier for switch_size_t, because it is different on different os's:
>> 
>> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_NOTICE, "Session %" SWITCH_SIZE_T_FMT " (%s) Ended\n",
>> 
>> you can use it like that, or optionally, if appropriate, cast the value to the right type when passing it to printf style commands.
>> 
> 
> Thanks Mike. I thought I had already done this but can find where it went.
> 
> I also need a function for big-endian platforms to convert native
> integer to little endian and the reverse. Those are simple to
> implement and I can do this, but I don't know where to put them
> (switch_utils.{h,c}?) and how to name them (switch_htol32,
> switch_ltoh32? or switch_native_to_little_endian,
> switch_to_little?...) ; or maybe they already exists?

we just use ntohl and htonl (see switch_rtp.c)

> 
> Patrick, his is solved on a2ceff1. Can you test again? For reference,
> this is MODSKINNY-5 on Jira.
> 
> Mathieu
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From steveu at coppice.org  Tue May 25 06:30:18 2010
From: steveu at coppice.org (Steve Underwood)
Date: Tue, 25 May 2010 21:30:18 +0800
Subject: [Freeswitch-users] Mod_skinny: Call for testing!
In-Reply-To: <5212488E-3EB9-43D9-85C9-99D809DA27D8@jerris.com>
References: <AANLkTik8hHRtOci8qTXmZVHOWvVxRg-OPm5zyRAMrWEe@mail.gmail.com>	<4BFA5A12.901@puzzled.xs4all.nl>	<F804849B-822F-4AC5-998E-577565BFE516@jerris.com>	<AANLkTinwptx6T23LN0IIM9nCVl19g-M7aYl2o8i7-eI1@mail.gmail.com>
	<5212488E-3EB9-43D9-85C9-99D809DA27D8@jerris.com>
Message-ID: <4BFBD0EA.1040407@coppice.org>

On 05/25/2010 08:57 PM, Michael Jerris wrote:
> On May 25, 2010, at 3:42 AM, Mathieu Parent wrote:
>
>    
>> On Tue, May 25, 2010 at 2:11 AM, Michael Jerris<mike at jerris.com>  wrote:
>>      
>>> Mathieu, we have a define for the printf specifier for switch_size_t, because it is different on different os's:
>>>
>>> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_NOTICE, "Session %" SWITCH_SIZE_T_FMT " (%s) Ended\n",
>>>
>>> you can use it like that, or optionally, if appropriate, cast the value to the right type when passing it to printf style commands.
>>>
>>>        
>> Thanks Mike. I thought I had already done this but can find where it went.
>>
>> I also need a function for big-endian platforms to convert native
>> integer to little endian and the reverse. Those are simple to
>> implement and I can do this, but I don't know where to put them
>> (switch_utils.{h,c}?) and how to name them (switch_htol32,
>> switch_ltoh32? or switch_native_to_little_endian,
>> switch_to_little?...) ; or maybe they already exists?
>>      
> we just use ntohl and htonl (see switch_rtp.c)
>    
Whjat happens when a long is 64 bits?
>> Patrick, his is solved on a2ceff1. Can you test again? For reference,
>> this is MODSKINNY-5 on Jira.
>>
>> Mathieu
>>      
Steve



From mike at yes.net.ua  Tue May 25 07:10:38 2010
From: mike at yes.net.ua (Mike Tkachuk)
Date: Tue, 25 May 2010 17:10:38 +0300
Subject: [Freeswitch-users] curl_xml
In-Reply-To: <AANLkTiltmx-qzAAgkQjUcCAGsAMp9_X573zbQic24pf7@mail.gmail.com>
References: <AANLkTiltmx-qzAAgkQjUcCAGsAMp9_X573zbQic24pf7@mail.gmail.com>
Message-ID: <1809326062.20100525171038@yes.net.ua>

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From anthony.minessale at gmail.com  Tue May 25 08:00:33 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 25 May 2010 10:00:33 -0500
Subject: [Freeswitch-users] sip_cid_type settings
In-Reply-To: <AABFAAFF-0CC1-46C6-A03E-E338906FB700@gmail.com>
References: <01991B335AA9418CB32018A477F5857F@ws4>
	<AABFAAFF-0CC1-46C6-A03E-E338906FB700@gmail.com>
Message-ID: <AANLkTimDN131YUiEYny9e5MgVNgLeIZB5wDAQKV3GgsP@mail.gmail.com>

well i was not sure so I looked in the example config and searched for dtmf
in under 10 seconds I was able to do the following from the build root.

cat conf/sip_profiles/internal.xml | grep dtmf
    <param name="dtmf-duration" value="2000"/>
    <!--<param name="dtmf-type" value="info"/>-->




2010/5/25 David Ponzone <david.ponzone at gmail.com>

> Frank,
>
> as far as I know, there is not.
> You can probably simulate that by using a custom variable in the gateway
> profile, like accountcode, and then matching that at the beginning of your
> dialplan to set the sip_cid_type accordingly.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 25/05/2010 ? 12:13, Frank @ Impact a ?crit :
>
> I can only find documentation on setting this value at the bridge level.
>
> Is there a way to control this setting in the gateway profile instead of
> having to do it in the bridge setup?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
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>
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pstn:+19193869900
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From phone.bytes at gmail.com  Tue May 25 08:03:30 2010
From: phone.bytes at gmail.com (Phone)
Date: Tue, 25 May 2010 09:03:30 -0600
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
In-Reply-To: <EC392AB972A94F8E8D8369C802771451@dell9400>
References: <490aedfdd7c8ac680bbe0d7f68c786de@mail.gmail.com>	<4BEB3DD4.9070807@gmail.com>
	<EC392AB972A94F8E8D8369C802771451@dell9400>
Message-ID: <4BFBE6C2.5000703@gmail.com>

Sorry to revive this already long thread...but I think the history is 
relevant as I am still stuck on the approach.

I came across the perl POE "POE::Filter::FSSocket" and thought I might 
learn and use it.  Thought it might be nice to help with the 
socket/threading issues...and avoid having to get down and dirty with 
them.  However, in looking closer it appeared that nothing had really 
happened on it for about 4 years...then I read a post  indicating that 
it was out of date and that Anthony's FreeSWITCH::CLient would be a 
better choice.  I have been able to try out the perl FreeSWITCH Client 
example on the wiki and  send out some calls using "bgapi"...but I am 
not sure how to handle the multiple threads.  Would I just deal with all 
events in a loop similar to what is in the wiki client example?

Now I am really confused as to how to tackle this.  Not an expert with 
any particular language...but it seems that scripting may be nice from a 
maintenance point of view??

Any further thoughts or examples to get me on my way?  I am really open 
as to the approach to take.

Thanks

So now I don't know what direction to go to get this started.
Jan Berger wrote:
> You don't need to pull a db these day's, you can use triggers to signal your
> application - a bit depending on what db you use and what scripting they
> offer beyond SQL. 
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Phone
> Sent: 13. mai 2010 01:46
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Questions on Building an application for
> FreeSWITCH
>
> Thanks for sharing your experience.
>
> We would like to get over to linux with this.
>
> I am still trying to get my head around how this generally works.  I 
> guess we would write a client to sit and run on the linux box where it 
> could periodically check the DB and then talk sockets to FS to make the 
> calls?
>
> Can you point me to examples of talking sockets?
>
>
>
>
>
> Kristian Kielhofner wrote:
>   
>> Glad to hear.
>>
>> I still say to build the app and logic completely separately and connect
>>     
> it
>   
>> to FreeSWITCH over the socket using ESL.
>>
>>
>> --
>> Kristian Kielhofner
>> http://blog.krisk.org
>>
>> ----- Original Message -----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> <freeswitch-users-bounces at lists.freeswitch.org>
>> To: freeswitch-users at lists.freeswitch.org
>> <freeswitch-users at lists.freeswitch.org>
>> Sent: Wed May 12 18:28:13 2010
>> Subject: Re: [Freeswitch-users] Questions on Building an application
>> forFreeSWITCHThis is legit.  We are not into harassing anyone.  There
>> are too many
>> people doing that already.
>>
>> We currently have this working on a Windows/Dialogic platform, but would
>> like to get it over to FreeSWITCH for a number of reasons.  It is always
>> nice to avoid spending alot of time developing using a method that you
>> later find has serious issues or limitations and then having to change
>> horses and start over.  I just thought I would tap on someone else's
>> wisdom in regards to a good way to talk to FS that works well.  I did
>> notice that the wiki talks like LUA is the preferred way to go.
>>
>> Thanks
>>
>>
>>
>> Michael Collins wrote:
>>   
>>     
>>> On Wed, May 12, 2010 at 2:50 PM, Kristian Kielhofner
>>> <kris at kriskinc.com <mailto:kris at kriskinc.com>> wrote:
>>>
>>>     Are you going to be calling my cell phone to ask if I owe $10,000
>>>     or more to
>>>     the IRS?
>>>
>>>     Seriously, the world does not need another robodialer to harass
>>>     people...
>>>
>>> Agreed, however there are legitimate use cases for this, like a
>>> retailer calling to say an order is in, or reminder calls sent out by
>>> a doctor's office. But yes, if it's dialing for dollars then I'm with
>>> you: Por favor! No mas!
>>> -MC
>>>
>>>
>>>
>>>     With that being said, write your app in whatever you want using
>>>     ESL and
>>>     connect to the socket to originate calls and listen for events.
>>>
>>>
>>>     --
>>>     Kristian Kielhofner
>>>     http://blog.krisk.org
>>>
>>>     ----- Original Message -----
>>>     From: freeswitch-users-bounces at lists.freeswitch.org
>>>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>
>>>     <freeswitch-users-bounces at lists.freeswitch.org
>>>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>>
>>>     To: freeswitch-users at lists.freeswitch.org
>>>     <mailto:freeswitch-users at lists.freeswitch.org>
>>>     <freeswitch-users at lists.freeswitch.org
>>>     <mailto:freeswitch-users at lists.freeswitch.org>>
>>>     Sent: Wed May 12 15:40:58 2010
>>>     Subject: [Freeswitch-users] Questions on Building an application
>>>     forFreeSWITCHI am looking for a boot in the right direction on the
>>>     following project.
>>>
>>>     I am looking to build an "application" with FS where it will
>>>     frequently
>>>     check a database and then make an outbound call when a record with
>>>     certain conditions is found in the DB.  It needs to determine if the
>>>     call was then answered by a machine/vm, busy, or answered by a
>>>       
> person,
>   
>>>     play a message and get a response back to be recorded in the db.
>>>      It may
>>>     find many calls that need to be made at the same time.  It will be
>>>     running on a PRI.
>>>
>>>     I am confused as to what approach to use to accomplish this.
>>>
>>>     Lua Script, Perl Script, Javascript, Phython, PHP, Ruby....etc.
>>>     Liverpie, ESL....etc.
>>>     Or a totally different approach.
>>>
>>>     I realize that there is not only one "correct" way to build this,
>>>     but I
>>>     am looking for advise from those who have done this type of thing
>>>     before.  I value the opinion of those who have gone down this road
>>>       
> and
>   
>>>     would be willing to share their thoughts on a recommended path to
>>>       
> take
>   
>>>     to accomplish this.
>>>
>>>     Thanks
>>>
>>>
>>>     _______________________________________________
>>>     FreeSWITCH-users mailing list
>>>     FreeSWITCH-users at lists.freeswitch.org
>>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     http://www.freeswitch.org
>>>
>>>     _______________________________________________
>>>     FreeSWITCH-users mailing list
>>>     FreeSWITCH-users at lists.freeswitch.org
>>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     http://www.freeswitch.org
>>>
>>>
>>> ------------------------------------------------------------------------
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>     
>>>       
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>   
>>     
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From mike at jerris.com  Tue May 25 08:08:59 2010
From: mike at jerris.com (Michael Jerris)
Date: Tue, 25 May 2010 11:08:59 -0400
Subject: [Freeswitch-users] Mod_skinny: Call for testing!
In-Reply-To: <4BFBD0EA.1040407@coppice.org>
References: <AANLkTik8hHRtOci8qTXmZVHOWvVxRg-OPm5zyRAMrWEe@mail.gmail.com>	<4BFA5A12.901@puzzled.xs4all.nl>	<F804849B-822F-4AC5-998E-577565BFE516@jerris.com>	<AANLkTinwptx6T23LN0IIM9nCVl19g-M7aYl2o8i7-eI1@mail.gmail.com>
	<5212488E-3EB9-43D9-85C9-99D809DA27D8@jerris.com>
	<4BFBD0EA.1040407@coppice.org>
Message-ID: <0BEFBE0A-7A5E-49AA-9827-98B7286769D8@jerris.com>


On May 25, 2010, at 9:30 AM, Steve Underwood wrote:

> On 05/25/2010 08:57 PM, Michael Jerris wrote:
>> On May 25, 2010, at 3:42 AM, Mathieu Parent wrote:
>> 
>> 
>>> On Tue, May 25, 2010 at 2:11 AM, Michael Jerris<mike at jerris.com>  wrote:
>>> 
>>>> Mathieu, we have a define for the printf specifier for switch_size_t, because it is different on different os's:
>>>> 
>>>> switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), SWITCH_LOG_NOTICE, "Session %" SWITCH_SIZE_T_FMT " (%s) Ended\n",
>>>> 
>>>> you can use it like that, or optionally, if appropriate, cast the value to the right type when passing it to printf style commands.
>>>> 
>>>> 
>>> Thanks Mike. I thought I had already done this but can find where it went.
>>> 
>>> I also need a function for big-endian platforms to convert native
>>> integer to little endian and the reverse. Those are simple to
>>> implement and I can do this, but I don't know where to put them
>>> (switch_utils.{h,c}?) and how to name them (switch_htol32,
>>> switch_ltoh32? or switch_native_to_little_endian,
>>> switch_to_little?...) ; or maybe they already exists?
>>> 
>> we just use ntohl and htonl (see switch_rtp.c)
>> 
> Whjat happens when a long is 64 bits?

We have not actually needed one so far I guess is the real answer, and yes, switch_utils.c is the right place for them.


>>> Patrick, his is solved on a2ceff1. Can you test again? For reference,
>>> this is MODSKINNY-5 on Jira.
>>> 
>>> Mathieu
>>> 
> Steve
> 



From anthony.minessale at gmail.com  Tue May 25 08:12:39 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 25 May 2010 10:12:39 -0500
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
In-Reply-To: <4BFBE6C2.5000703@gmail.com>
References: <490aedfdd7c8ac680bbe0d7f68c786de@mail.gmail.com>
	<4BEB3DD4.9070807@gmail.com>
	<EC392AB972A94F8E8D8369C802771451@dell9400>
	<4BFBE6C2.5000703@gmail.com>
Message-ID: <AANLkTilsHn9znajzF0Jj-QDu4je9QW7CeymuBYE0SyKa@mail.gmail.com>

Where did you read all of that because it's out of date.
All of the above is deprecated in favor of ESL and the perl module for ESL.
That should have been much easier to find than the stuff you are referencing
since ESL was even mentioned
in this very thread.

What threading issues are you having in perl?  Are you actually going to use
perl for a threaded app?


On Tue, May 25, 2010 at 10:03 AM, Phone <phone.bytes at gmail.com> wrote:

> Sorry to revive this already long thread...but I think the history is
> relevant as I am still stuck on the approach.
>
> I came across the perl POE "POE::Filter::FSSocket" and thought I might
> learn and use it.  Thought it might be nice to help with the
> socket/threading issues...and avoid having to get down and dirty with
> them.  However, in looking closer it appeared that nothing had really
> happened on it for about 4 years...then I read a post  indicating that
> it was out of date and that Anthony's FreeSWITCH::CLient would be a
> better choice.  I have been able to try out the perl FreeSWITCH Client
> example on the wiki and  send out some calls using "bgapi"...but I am
> not sure how to handle the multiple threads.  Would I just deal with all
> events in a loop similar to what is in the wiki client example?
>
> Now I am really confused as to how to tackle this.  Not an expert with
> any particular language...but it seems that scripting may be nice from a
> maintenance point of view??
>
> Any further thoughts or examples to get me on my way?  I am really open
> as to the approach to take.
>
> Thanks
>
> So now I don't know what direction to go to get this started.
> Jan Berger wrote:
> > You don't need to pull a db these day's, you can use triggers to signal
> your
> > application - a bit depending on what db you use and what scripting they
> > offer beyond SQL.
> >
> > -----Original Message-----
> > From: freeswitch-users-bounces at lists.freeswitch.org
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Phone
> > Sent: 13. mai 2010 01:46
> > To: freeswitch-users at lists.freeswitch.org
> > Subject: Re: [Freeswitch-users] Questions on Building an application for
> > FreeSWITCH
> >
> > Thanks for sharing your experience.
> >
> > We would like to get over to linux with this.
> >
> > I am still trying to get my head around how this generally works.  I
> > guess we would write a client to sit and run on the linux box where it
> > could periodically check the DB and then talk sockets to FS to make the
> > calls?
> >
> > Can you point me to examples of talking sockets?
> >
> >
> >
> >
> >
> > Kristian Kielhofner wrote:
> >
> >> Glad to hear.
> >>
> >> I still say to build the app and logic completely separately and connect
> >>
> > it
> >
> >> to FreeSWITCH over the socket using ESL.
> >>
> >>
> >> --
> >> Kristian Kielhofner
> >> http://blog.krisk.org
> >>
> >> ----- Original Message -----
> >> From: freeswitch-users-bounces at lists.freeswitch.org
> >> <freeswitch-users-bounces at lists.freeswitch.org>
> >> To: freeswitch-users at lists.freeswitch.org
> >> <freeswitch-users at lists.freeswitch.org>
> >> Sent: Wed May 12 18:28:13 2010
> >> Subject: Re: [Freeswitch-users] Questions on Building an application
> >> forFreeSWITCHThis is legit.  We are not into harassing anyone.  There
> >> are too many
> >> people doing that already.
> >>
> >> We currently have this working on a Windows/Dialogic platform, but would
> >> like to get it over to FreeSWITCH for a number of reasons.  It is always
> >> nice to avoid spending alot of time developing using a method that you
> >> later find has serious issues or limitations and then having to change
> >> horses and start over.  I just thought I would tap on someone else's
> >> wisdom in regards to a good way to talk to FS that works well.  I did
> >> notice that the wiki talks like LUA is the preferred way to go.
> >>
> >> Thanks
> >>
> >>
> >>
> >> Michael Collins wrote:
> >>
> >>
> >>> On Wed, May 12, 2010 at 2:50 PM, Kristian Kielhofner
> >>> <kris at kriskinc.com <mailto:kris at kriskinc.com>> wrote:
> >>>
> >>>     Are you going to be calling my cell phone to ask if I owe $10,000
> >>>     or more to
> >>>     the IRS?
> >>>
> >>>     Seriously, the world does not need another robodialer to harass
> >>>     people...
> >>>
> >>> Agreed, however there are legitimate use cases for this, like a
> >>> retailer calling to say an order is in, or reminder calls sent out by
> >>> a doctor's office. But yes, if it's dialing for dollars then I'm with
> >>> you: Por favor! No mas!
> >>> -MC
> >>>
> >>>
> >>>
> >>>     With that being said, write your app in whatever you want using
> >>>     ESL and
> >>>     connect to the socket to originate calls and listen for events.
> >>>
> >>>
> >>>     --
> >>>     Kristian Kielhofner
> >>>     http://blog.krisk.org
> >>>
> >>>     ----- Original Message -----
> >>>     From: freeswitch-users-bounces at lists.freeswitch.org
> >>>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>
> >>>     <freeswitch-users-bounces at lists.freeswitch.org
> >>>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>>
> >>>     To: freeswitch-users at lists.freeswitch.org
> >>>     <mailto:freeswitch-users at lists.freeswitch.org>
> >>>     <freeswitch-users at lists.freeswitch.org
> >>>     <mailto:freeswitch-users at lists.freeswitch.org>>
> >>>     Sent: Wed May 12 15:40:58 2010
> >>>     Subject: [Freeswitch-users] Questions on Building an application
> >>>     forFreeSWITCHI am looking for a boot in the right direction on the
> >>>     following project.
> >>>
> >>>     I am looking to build an "application" with FS where it will
> >>>     frequently
> >>>     check a database and then make an outbound call when a record with
> >>>     certain conditions is found in the DB.  It needs to determine if
> the
> >>>     call was then answered by a machine/vm, busy, or answered by a
> >>>
> > person,
> >
> >>>     play a message and get a response back to be recorded in the db.
> >>>      It may
> >>>     find many calls that need to be made at the same time.  It will be
> >>>     running on a PRI.
> >>>
> >>>     I am confused as to what approach to use to accomplish this.
> >>>
> >>>     Lua Script, Perl Script, Javascript, Phython, PHP, Ruby....etc.
> >>>     Liverpie, ESL....etc.
> >>>     Or a totally different approach.
> >>>
> >>>     I realize that there is not only one "correct" way to build this,
> >>>     but I
> >>>     am looking for advise from those who have done this type of thing
> >>>     before.  I value the opinion of those who have gone down this road
> >>>
> > and
> >
> >>>     would be willing to share their thoughts on a recommended path to
> >>>
> > take
> >
> >>>     to accomplish this.
> >>>
> >>>     Thanks
> >>>
> >>>
> >>>     _______________________________________________
> >>>     FreeSWITCH-users mailing list
> >>>     FreeSWITCH-users at lists.freeswitch.org
> >>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>     http://www.freeswitch.org
> >>>
> >>>     _______________________________________________
> >>>     FreeSWITCH-users mailing list
> >>>     FreeSWITCH-users at lists.freeswitch.org
> >>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>     http://www.freeswitch.org
> >>>
> >>>
> >>>
> ------------------------------------------------------------------------
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>>
> >>>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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FreeSWITCH Developer Conference
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From phone.bytes at gmail.com  Tue May 25 08:40:44 2010
From: phone.bytes at gmail.com (Phone)
Date: Tue, 25 May 2010 09:40:44 -0600
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
In-Reply-To: <AANLkTilsHn9znajzF0Jj-QDu4je9QW7CeymuBYE0SyKa@mail.gmail.com>
References: <490aedfdd7c8ac680bbe0d7f68c786de@mail.gmail.com>	<4BEB3DD4.9070807@gmail.com>	<EC392AB972A94F8E8D8369C802771451@dell9400>	<4BFBE6C2.5000703@gmail.com>
	<AANLkTilsHn9znajzF0Jj-QDu4je9QW7CeymuBYE0SyKa@mail.gmail.com>
Message-ID: <4BFBEF7C.9020107@gmail.com>

As I mentioned...I am a bit confused now...I guess your reply confirms 
that...sorry...I have been scouring the web to learn what route to 
take...I guess I should stay closer to the source.   I don't mean to ask 
stupid questions...I am just trying to build an app with this great tool 
FreeSWITCH...and it is new to me.

My understanding is that I will need to use a threaded app. I am checkng 
a DB for pending calls, placing multiple calls 
simultaneously....watching for them to be answered and then take action 
based upon dtmf or other response.  Am I correct that I will use 
threading and events for this?

Sounds like perl for a threaded app is not a good idea. I have not 
really got very far threading in perl. I was just wondering if it was a 
viable solution to build from the perl client example by expanding the 
event loop there or if I was way off and should be going a different 
direction.  What do you suggest?

Thanks again for your help...I will ask smarter questions when I get going.



Anthony Minessale wrote:
> Where did you read all of that because it's out of date.
> All of the above is deprecated in favor of ESL and the perl module for 
> ESL.
> That should have been much easier to find than the stuff you are 
> referencing since ESL was even mentioned
> in this very thread.
>
> What threading issues are you having in perl?  Are you actually 
> going to use perl for a threaded app?
>
>
> On Tue, May 25, 2010 at 10:03 AM, Phone <phone.bytes at gmail.com 
> <mailto:phone.bytes at gmail.com>> wrote:
>
>     Sorry to revive this already long thread...but I think the history is
>     relevant as I am still stuck on the approach.
>
>     I came across the perl POE "POE::Filter::FSSocket" and thought I might
>     learn and use it.  Thought it might be nice to help with the
>     socket/threading issues...and avoid having to get down and dirty with
>     them.  However, in looking closer it appeared that nothing had really
>     happened on it for about 4 years...then I read a post  indicating that
>     it was out of date and that Anthony's FreeSWITCH::CLient would be a
>     better choice.  I have been able to try out the perl FreeSWITCH Client
>     example on the wiki and  send out some calls using "bgapi"...but I am
>     not sure how to handle the multiple threads.  Would I just deal
>     with all
>     events in a loop similar to what is in the wiki client example?
>
>     Now I am really confused as to how to tackle this.  Not an expert with
>     any particular language...but it seems that scripting may be nice
>     from a
>     maintenance point of view??
>
>     Any further thoughts or examples to get me on my way?  I am really
>     open
>     as to the approach to take.
>
>     Thanks
>
>     So now I don't know what direction to go to get this started.
>     Jan Berger wrote:
>     > You don't need to pull a db these day's, you can use triggers to
>     signal your
>     > application - a bit depending on what db you use and what
>     scripting they
>     > offer beyond SQL.
>     >
>     > -----Original Message-----
>     > From: freeswitch-users-bounces at lists.freeswitch.org
>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>
>     > [mailto:freeswitch-users-bounces at lists.freeswitch.org
>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf
>     Of Phone
>     > Sent: 13. mai 2010 01:46
>     > To: freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>
>     > Subject: Re: [Freeswitch-users] Questions on Building an
>     application for
>     > FreeSWITCH
>     >
>     > Thanks for sharing your experience.
>     >
>     > We would like to get over to linux with this.
>     >
>     > I am still trying to get my head around how this generally works.  I
>     > guess we would write a client to sit and run on the linux box
>     where it
>     > could periodically check the DB and then talk sockets to FS to
>     make the
>     > calls?
>     >
>     > Can you point me to examples of talking sockets?
>     >
>     >
>     >
>     >
>     >
>     > Kristian Kielhofner wrote:
>     >
>     >> Glad to hear.
>     >>
>     >> I still say to build the app and logic completely separately
>     and connect
>     >>
>     > it
>     >
>     >> to FreeSWITCH over the socket using ESL.
>     >>
>     >>
>     >> --
>     >> Kristian Kielhofner
>     >> http://blog.krisk.org
>     >>
>     >> ----- Original Message -----
>     >> From: freeswitch-users-bounces at lists.freeswitch.org
>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>
>     >> <freeswitch-users-bounces at lists.freeswitch.org
>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>>
>     >> To: freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>
>     >> <freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>>
>     >> Sent: Wed May 12 18:28:13 2010
>     >> Subject: Re: [Freeswitch-users] Questions on Building an
>     application
>     >> forFreeSWITCHThis is legit.  We are not into harassing anyone.
>      There
>     >> are too many
>     >> people doing that already.
>     >>
>     >> We currently have this working on a Windows/Dialogic platform,
>     but would
>     >> like to get it over to FreeSWITCH for a number of reasons.  It
>     is always
>     >> nice to avoid spending alot of time developing using a method
>     that you
>     >> later find has serious issues or limitations and then having to
>     change
>     >> horses and start over.  I just thought I would tap on someone
>     else's
>     >> wisdom in regards to a good way to talk to FS that works well.
>      I did
>     >> notice that the wiki talks like LUA is the preferred way to go.
>     >>
>     >> Thanks
>     >>
>     >>
>     >>
>     >> Michael Collins wrote:
>     >>
>     >>
>     >>> On Wed, May 12, 2010 at 2:50 PM, Kristian Kielhofner
>     >>> <kris at kriskinc.com <mailto:kris at kriskinc.com>
>     <mailto:kris at kriskinc.com <mailto:kris at kriskinc.com>>> wrote:
>     >>>
>     >>>     Are you going to be calling my cell phone to ask if I owe
>     $10,000
>     >>>     or more to
>     >>>     the IRS?
>     >>>
>     >>>     Seriously, the world does not need another robodialer to
>     harass
>     >>>     people...
>     >>>
>     >>> Agreed, however there are legitimate use cases for this, like a
>     >>> retailer calling to say an order is in, or reminder calls sent
>     out by
>     >>> a doctor's office. But yes, if it's dialing for dollars then
>     I'm with
>     >>> you: Por favor! No mas!
>     >>> -MC
>     >>>
>     >>>
>     >>>
>     >>>     With that being said, write your app in whatever you want
>     using
>     >>>     ESL and
>     >>>     connect to the socket to originate calls and listen for
>     events.
>     >>>
>     >>>
>     >>>     --
>     >>>     Kristian Kielhofner
>     >>>     http://blog.krisk.org
>     >>>
>     >>>     ----- Original Message -----
>     >>>     From: freeswitch-users-bounces at lists.freeswitch.org
>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>
>     >>>     <mailto:freeswitch-users-bounces at lists.freeswitch.org
>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>>
>     >>>     <freeswitch-users-bounces at lists.freeswitch.org
>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>
>     >>>     <mailto:freeswitch-users-bounces at lists.freeswitch.org
>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>>>
>     >>>     To: freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>
>     >>>     <mailto:freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>>
>     >>>     <freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>
>     >>>     <mailto:freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>>>
>     >>>     Sent: Wed May 12 15:40:58 2010
>     >>>     Subject: [Freeswitch-users] Questions on Building an
>     application
>     >>>     forFreeSWITCHI am looking for a boot in the right
>     direction on the
>     >>>     following project.
>     >>>
>     >>>     I am looking to build an "application" with FS where it will
>     >>>     frequently
>     >>>     check a database and then make an outbound call when a
>     record with
>     >>>     certain conditions is found in the DB.  It needs to
>     determine if the
>     >>>     call was then answered by a machine/vm, busy, or answered by a
>     >>>
>     > person,
>     >
>     >>>     play a message and get a response back to be recorded in
>     the db.
>     >>>      It may
>     >>>     find many calls that need to be made at the same time.  It
>     will be
>     >>>     running on a PRI.
>     >>>
>     >>>     I am confused as to what approach to use to accomplish this.
>     >>>
>     >>>     Lua Script, Perl Script, Javascript, Phython, PHP,
>     Ruby....etc.
>     >>>     Liverpie, ESL....etc.
>     >>>     Or a totally different approach.
>     >>>
>     >>>     I realize that there is not only one "correct" way to
>     build this,
>     >>>     but I
>     >>>     am looking for advise from those who have done this type
>     of thing
>     >>>     before.  I value the opinion of those who have gone down
>     this road
>     >>>
>     > and
>     >
>     >>>     would be willing to share their thoughts on a recommended
>     path to
>     >>>
>     > take
>     >
>     >>>     to accomplish this.
>     >>>
>     >>>     Thanks
>     >>>
>     >>>
>     >>>     _______________________________________________
>     >>>     FreeSWITCH-users mailing list
>     >>>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >>>     <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>     >>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >>>
>     >>>
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >>>     http://www.freeswitch.org
>     >>>
>     >>>     _______________________________________________
>     >>>     FreeSWITCH-users mailing list
>     >>>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >>>     <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>     >>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >>>
>     >>>
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >>>     http://www.freeswitch.org
>     >>>
>     >>>
>     >>>
>     ------------------------------------------------------------------------
>     >>>
>     >>> _______________________________________________
>     >>> FreeSWITCH-users mailing list
>     >>> FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >>>
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >>> http://www.freeswitch.org
>     >>>
>     >>>
>     >>>
>     >> _______________________________________________
>     >> FreeSWITCH-users mailing list
>     >> FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >>
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >> http://www.freeswitch.org
>     >>
>     >> _______________________________________________
>     >> FreeSWITCH-users mailing list
>     >> FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >>
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >> http://www.freeswitch.org
>     >>
>     >>
>     >
>     >
>     > _______________________________________________
>     > FreeSWITCH-users mailing list
>     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > http://www.freeswitch.org
>     >
>     >
>     > _______________________________________________
>     > FreeSWITCH-users mailing list
>     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > http://www.freeswitch.org
>     >
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From infos at madovsky.org  Tue May 25 09:57:24 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 25 May 2010 12:57:24 -0400
Subject: [Freeswitch-users] get a freeswitch var from PHP
Message-ID: <A9B2CF8578BA4764A5313248EDCE13BE@MOBILEE1705>

Hi all,

I set a special var in my dialplan before the bridge that I'd like to get from PHP
before the call is done (it's a var that define which trunk choosed from distributor module).
Which is the elegantest way to do it ?

Best Regards

Franck
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From david.ponzone at gmail.com  Tue May 25 10:27:43 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Tue, 25 May 2010 19:27:43 +0200
Subject: [Freeswitch-users] get a freeswitch var from PHP
In-Reply-To: <A9B2CF8578BA4764A5313248EDCE13BE@MOBILEE1705>
References: <A9B2CF8578BA4764A5313248EDCE13BE@MOBILEE1705>
Message-ID: <E26DC27E-D1C7-483F-98BA-325A679E2572@gmail.com>

 From PHP ?
You mean fro PHP/ESL ?

Call done ?
You meant call sent ou call finished ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 25/05/2010 ? 18:57, Madovsky a ?crit :

> Hi all,
>
> I set a special var in my dialplan before the bridge that I'd like  
> to get from PHP
> before the call is done (it's a var that define which trunk choosed  
> from distributor module).
> Which is the elegantest way to do it ?
>
> Best Regards
>
> Franck
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From infos at madovsky.org  Tue May 25 10:40:01 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 25 May 2010 13:40:01 -0400
Subject: [Freeswitch-users] get a freeswitch var from PHP
References: <A9B2CF8578BA4764A5313248EDCE13BE@MOBILEE1705>
	<E26DC27E-D1C7-483F-98BA-325A679E2572@gmail.com>
Message-ID: <16132F5DC2A44121B3EF6A6C8A3B55F8@MOBILEE1705>

yes why not.
or from PHP by a exec fs_cli call if possible ?
maybe I should reformulate my question.. from fs_cli ? :)
  ----- Original Message ----- 
  From: David Ponzone 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, May 25, 2010 1:27 PM
  Subject: Re: [Freeswitch-users] get a freeswitch var from PHP


  From PHP ?
  You mean fro PHP/ESL ?


  Call done ?
  You meant call sent ou call finished ?


  David Ponzone  Direction Technique
  email: david.ponzone at ipeva.fr
  tel:      01 74 03 18 97
  gsm:   06 66 98 76 34


  Service Client IPeva
  tel:      0811 46 26 26
  www.ipeva.fr  -   www.ipeva-studio.com


  Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.








  Le 25/05/2010 ? 18:57, Madovsky a ?crit :


    Hi all,

    I set a special var in my dialplan before the bridge that I'd like to get from PHP
    before the call is done (it's a var that define which trunk choosed from distributor module).
    Which is the elegantest way to do it ?

    Best Regards

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





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From infos at madovsky.org  Tue May 25 10:42:11 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 25 May 2010 13:42:11 -0400
Subject: [Freeswitch-users] G723 codec
Message-ID: <BE2ABE1897F946478254BE3F193F0581@MOBILEE1705>

Hi all,

is there a chance that in future there will be a mod_g723 transcoder ?

Thanks

Franck
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From brian at freeswitch.org  Tue May 25 10:42:38 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 25 May 2010 12:42:38 -0500
Subject: [Freeswitch-users] get a freeswitch var from PHP
In-Reply-To: <16132F5DC2A44121B3EF6A6C8A3B55F8@MOBILEE1705>
References: <A9B2CF8578BA4764A5313248EDCE13BE@MOBILEE1705>
	<E26DC27E-D1C7-483F-98BA-325A679E2572@gmail.com>
	<16132F5DC2A44121B3EF6A6C8A3B55F8@MOBILEE1705>
Message-ID: <3D453DAA-610E-43DF-BADD-C81AB5861C8C@freeswitch.org>

fs_cli -x

/b

On May 25, 2010, at 12:40 PM, Madovsky wrote:

> yes why not.
> or from PHP by a exec fs_cli call if possible ?
> maybe I should reformulate my question.. from fs_cli ? :)

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From brian at freeswitch.org  Tue May 25 10:46:41 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 25 May 2010 12:46:41 -0500
Subject: [Freeswitch-users] G723 codec
In-Reply-To: <BE2ABE1897F946478254BE3F193F0581@MOBILEE1705>
References: <BE2ABE1897F946478254BE3F193F0581@MOBILEE1705>
Message-ID: <B45AB3AB-0C94-42AE-99C5-81B65ED6EC18@freeswitch.org>

No, the patents on that are so close to expiring anyway its not worth it to do the licensing for it...

See when a codec nears the end these slimy patent pools like to crank up the prices to squeeze every lsat drop out of it.

Anyone recall the date it expires?

/b

On May 25, 2010, at 12:42 PM, Madovsky wrote:

> Hi all,
>  
> is there a chance that in future there will be a mod_g723 transcoder ?
>  
> Thanks
>  
> Franck

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From brian at freeswitch.org  Tue May 25 10:47:05 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 25 May 2010 12:47:05 -0500
Subject: [Freeswitch-users] G723 codec
In-Reply-To: <BE2ABE1897F946478254BE3F193F0581@MOBILEE1705>
References: <BE2ABE1897F946478254BE3F193F0581@MOBILEE1705>
Message-ID: <00FB6751-3766-4A08-AAFE-4F7E0330472F@freeswitch.org>

The last patent will expire in 2014.

/b

On May 25, 2010, at 12:42 PM, Madovsky wrote:

> Hi all,
>  
> is there a chance that in future there will be a mod_g723 transcoder ?
>  
> Thanks
>  
> Franck
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From infos at madovsky.org  Tue May 25 10:57:08 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 25 May 2010 13:57:08 -0400
Subject: [Freeswitch-users] vars set in external trunk
Message-ID: <7DE4725D17DE4359B6B6A37D48B1DBB2@MOBILEE1705>

Hi all,

should I us $${var} or ${var} to call a var from a dialplan set in a external sip profile ?

Thanks
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From davis.erwin at gmail.com  Tue May 25 10:59:41 2010
From: davis.erwin at gmail.com (Erwin Davis)
Date: Tue, 25 May 2010 10:59:41 -0700
Subject: [Freeswitch-users] curl_xml
In-Reply-To: <1809326062.20100525171038@yes.net.ua>
References: <AANLkTiltmx-qzAAgkQjUcCAGsAMp9_X573zbQic24pf7@mail.gmail.com>
	<1809326062.20100525171038@yes.net.ua>
Message-ID: <AANLkTikcwFu_GYpm_hlFHc3TZzDO5-8vjNlslKpQgBHA@mail.gmail.com>

yes, I fixed it with the new xml layout as suggested. Thank you, everyone
who was trying to help and lead me to the right direction. You are awesome.

Hi, Anthony, my apologies. But this is the happy ending of this thread. I
promise. -:)

On Tue, May 25, 2010 at 7:10 AM, Mike Tkachuk <mike at yes.net.ua> wrote:

>  Hello Erwin,
>
>  Also don't forget to set "Content-Type" header to "text/xml"
>
>
> Wednesday, May 19, 2010 5:21:16 PM, you wrote:
>
>
>  Hi, I am trying to use curl_xml to dynamically send back the response for
> directory query from a webserver.
> But when the FS received the response from the web server, it kept sending
> 403 forbidden response to the sip client (x-lite).
> I post the FS log from  the console in
> http://pastebin.freeswitch.org/13007 and the XML response log in
> http://pastebin.freeswitch.org/13007. Thanks for your help,
>
>
>
> --
> Mike Tkachuk
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Tue May 25 11:00:18 2010
From: brian at freeswitch.org (Brian West)
Date: Tue, 25 May 2010 13:00:18 -0500
Subject: [Freeswitch-users] vars set in external trunk
In-Reply-To: <7DE4725D17DE4359B6B6A37D48B1DBB2@MOBILEE1705>
References: <7DE4725D17DE4359B6B6A37D48B1DBB2@MOBILEE1705>
Message-ID: <5567677B-14AF-49B5-B889-0E720FB3F159@freeswitch.org>

The FAQ covers this.

http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ#Q:_What_is_the_difference_between_using_a_.24.7Bvar.7D_and_.24.24.7Bvar.7D_in_the_configuration_files.3F

/b

On May 25, 2010, at 12:57 PM, Madovsky wrote:

> Hi all,
>  
> should I us $${var} or ${var} to call a var from a dialplan set in a external sip profile ?
>  
> Thanks

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From steveu at coppice.org  Tue May 25 11:07:01 2010
From: steveu at coppice.org (Steve Underwood)
Date: Wed, 26 May 2010 02:07:01 +0800
Subject: [Freeswitch-users] G723 codec
In-Reply-To: <BE2ABE1897F946478254BE3F193F0581@MOBILEE1705>
References: <BE2ABE1897F946478254BE3F193F0581@MOBILEE1705>
Message-ID: <4BFC11C5.9080601@coppice.org>

On 05/26/2010 01:42 AM, Madovsky wrote:
> Hi all,
> is there a chance that in future there will be a mod_g723 transcoder ?
>
I assume you mean G.723.1. It is very difficult to licence the patents 
related to G.723.1 unless you are a huge user. G.729 has a well defined 
scheme for licencing the relevant patents on an incremental basis, 
starting at a modest numbers of channels. G.723.1 is quite problematic. 
For that reason it is unlikely that G.723.1 will be available as a host 
software transcoder module any time soon. There are, however, hardware 
card options which can provide this codec.

Steve



From msc at freeswitch.org  Tue May 25 11:11:37 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 25 May 2010 11:11:37 -0700
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
In-Reply-To: <4BFBEF7C.9020107@gmail.com>
References: <490aedfdd7c8ac680bbe0d7f68c786de@mail.gmail.com>
	<4BEB3DD4.9070807@gmail.com>
	<EC392AB972A94F8E8D8369C802771451@dell9400>
	<4BFBE6C2.5000703@gmail.com>
	<AANLkTilsHn9znajzF0Jj-QDu4je9QW7CeymuBYE0SyKa@mail.gmail.com>
	<4BFBEF7C.9020107@gmail.com>
Message-ID: <AANLkTim-7gW8JN7qcsUAFP5h0Ckge57gdL-sVSrrfEbr@mail.gmail.com>

On Tue, May 25, 2010 at 8:40 AM, Phone <phone.bytes at gmail.com> wrote:

> As I mentioned...I am a bit confused now...I guess your reply confirms
> that...sorry...I have been scouring the web to learn what route to
> take...I guess I should stay closer to the source.   I don't mean to ask
> stupid questions...I am just trying to build an app with this great tool
> FreeSWITCH...and it is new to me.
>
It's okay. You just happened to have picked an advanced project for someone
knew to FreeSWITCH and also relatively new to programming.


>
> My understanding is that I will need to use a threaded app. I am checkng
> a DB for pending calls, placing multiple calls
> simultaneously....watching for them to be answered and then take action
> based upon dtmf or other response.  Am I correct that I will use
> threading and events for this?
>
The term "threaded" might be misleading. Yes, a threaded app would probably
work, but depending on your needs it isn't strictly necessary. Perl's POE is
not a "threaded" app but it is an advanced event loop that allows many
things to be happening at once. (Strictly speaking it just quickly switches
between tasks.) You just need things to be asynchronous, aka "non-blocking"
so that your program doesn't do stuff like submit an originate and sit there
for 20 seconds doing nothing else while it waits for the originate to
succeed or fail. (That's why you use "bgapi" like I mentioned before.)


>
> Sounds like perl for a threaded app is not a good idea. I have not
> really got very far threading in perl. I was just wondering if it was a
> viable solution to build from the perl client example by expanding the
> event loop there or if I was way off and should be going a different
> direction.  What do you suggest?
>
We suggest you use ESL and whichever programming language you are most
comfortable with, assuming that you are comfortable with any of those that
support ESL. :)  Perl has POE. Python was Twisted (and the fiorix
framework). C has, of course, the ESL itself.


>
> Thanks again for your help...I will ask smarter questions when I get going.
>
My advice to you is to get out the pencil and paper and sketch this out if
you haven't already done so. Last year at ClueCon Tony talked about the
thinking process that he went through prior to coding a single line of
FreeSWITCH. He went of to a little Zen place and thought things through.
It's much easier to fix things that are broken before you start coding. The
next step for you is to familiarize yourself with the various tools at your
disposal. You have FreeSWITCH, ESL, a scripting language, a database, and
probably some sort of UI to interface with that database. Frankly, FS is the
least of your problems. Oh, and did I mention CDRs? :) You have to do
something with the call results, no? And will people want to know what
happened to all of those calls that were made? (Reporting.) Like Tony says,
make your mistakes while you are thinking things through - they will be much
easier to fix if you haven't started coding yet.

You've taken on an ambitious project. We are not trying to discourage you
from doing it - we just want to make sure that you know what you're getting
into.

-MC
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From msc at freeswitch.org  Tue May 25 11:23:26 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 25 May 2010 11:23:26 -0700
Subject: [Freeswitch-users] get a freeswitch var from PHP
In-Reply-To: <3D453DAA-610E-43DF-BADD-C81AB5861C8C@freeswitch.org>
References: <A9B2CF8578BA4764A5313248EDCE13BE@MOBILEE1705>
	<E26DC27E-D1C7-483F-98BA-325A679E2572@gmail.com>
	<16132F5DC2A44121B3EF6A6C8A3B55F8@MOBILEE1705>
	<3D453DAA-610E-43DF-BADD-C81AB5861C8C@freeswitch.org>
Message-ID: <AANLkTilFIguFmRDUXwWCqIffCOfqTyDHdm294_3r0x7T@mail.gmail.com>

On Tue, May 25, 2010 at 10:42 AM, Brian West <brian at freeswitch.org> wrote:

> fs_cli -x
>
>
So if you know the uuid of the call this do this:

fs_cli -x "uuid_getvar <uuid> <varname>"

Here's an example. From fs_cli I did this:
originate {my_test_var=abc123xyz}loopback/9999 9999

That creates two channels, one of which has 'my_test_var' set to
'abc123xyz'.
show channels gives me the uuid: dc773eec-4dfe-4e8d-841d-ccfb4848658a

At the Linux command line:

root at freeswitch1 [Tue May 25 11:21 AM]  /usr/local/freeswitch
<25>:fs_cli -x "uuid_getvar dc773eec-4dfe-4e8d-841d-ccfb4848658a
my_test_var"
abc123xyz

>From ESL you have the $con object where you can do a uuid_getvar as well:
$var_contents = $con->api('uuid_getvar', "$uuid $var_name");  # Perl example

Have fun.
-MC
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From msc at freeswitch.org  Tue May 25 11:24:34 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 25 May 2010 11:24:34 -0700
Subject: [Freeswitch-users] curl_xml
In-Reply-To: <AANLkTikcwFu_GYpm_hlFHc3TZzDO5-8vjNlslKpQgBHA@mail.gmail.com>
References: <AANLkTiltmx-qzAAgkQjUcCAGsAMp9_X573zbQic24pf7@mail.gmail.com>
	<1809326062.20100525171038@yes.net.ua>
	<AANLkTikcwFu_GYpm_hlFHc3TZzDO5-8vjNlslKpQgBHA@mail.gmail.com>
Message-ID: <AANLkTilNgIfBE_I8B2_pa4f5RODgX8mcU9jUg0_-tHMF@mail.gmail.com>

On Tue, May 25, 2010 at 10:59 AM, Erwin Davis <davis.erwin at gmail.com> wrote:

> yes, I fixed it with the new xml layout as suggested. Thank you, everyone
> who was trying to help and lead me to the right direction. You are awesome.
>
> Hi, Anthony, my apologies. But this is the happy ending of this thread. I
> promise. -:)
>

I can hear the "Hallelujah" chorus playing in the background! :P
-MC
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From jan.berger at video24.no  Tue May 25 11:38:54 2010
From: jan.berger at video24.no (Jan Berger)
Date: Tue, 25 May 2010 20:38:54 +0200
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
In-Reply-To: <4BFBEF7C.9020107@gmail.com>
References: <490aedfdd7c8ac680bbe0d7f68c786de@mail.gmail.com>
	<4BEB3DD4.9070807@gmail.com>	<EC392AB972A94F8E8D8369C802771451@dell9400>
	<4BFBE6C2.5000703@gmail.com>
	<AANLkTilsHn9znajzF0Jj-QDu4je9QW7CeymuBYE0SyKa@mail.gmail.com>
	<4BFBEF7C.9020107@gmail.com>
Message-ID: <9250A9FF05DB4B879DEF6ECE8E3FFAC1@dell9400>

Actually - before you get "to smart" - may I suggest that you start writing
- or improving - the getting started sections of the doc. Address the areas
where you struggle and let others benefit from your work. 

I have been through similar issues myself - FS is one of the easier projects
to work with once you get under the hood, but you basically need to evolve
to the level where you read the source code.

Jan

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Phone
Sent: 25. mai 2010 17:41
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Questions on Building an application for
FreeSWITCH

As I mentioned...I am a bit confused now...I guess your reply confirms 
that...sorry...I have been scouring the web to learn what route to 
take...I guess I should stay closer to the source.   I don't mean to ask 
stupid questions...I am just trying to build an app with this great tool 
FreeSWITCH...and it is new to me.

My understanding is that I will need to use a threaded app. I am checkng 
a DB for pending calls, placing multiple calls 
simultaneously....watching for them to be answered and then take action 
based upon dtmf or other response.  Am I correct that I will use 
threading and events for this?

Sounds like perl for a threaded app is not a good idea. I have not 
really got very far threading in perl. I was just wondering if it was a 
viable solution to build from the perl client example by expanding the 
event loop there or if I was way off and should be going a different 
direction.  What do you suggest?

Thanks again for your help...I will ask smarter questions when I get going.



Anthony Minessale wrote:
> Where did you read all of that because it's out of date.
> All of the above is deprecated in favor of ESL and the perl module for 
> ESL.
> That should have been much easier to find than the stuff you are 
> referencing since ESL was even mentioned
> in this very thread.
>
> What threading issues are you having in perl?  Are you actually 
> going to use perl for a threaded app?
>
>
> On Tue, May 25, 2010 at 10:03 AM, Phone <phone.bytes at gmail.com 
> <mailto:phone.bytes at gmail.com>> wrote:
>
>     Sorry to revive this already long thread...but I think the history is
>     relevant as I am still stuck on the approach.
>
>     I came across the perl POE "POE::Filter::FSSocket" and thought I might
>     learn and use it.  Thought it might be nice to help with the
>     socket/threading issues...and avoid having to get down and dirty with
>     them.  However, in looking closer it appeared that nothing had really
>     happened on it for about 4 years...then I read a post  indicating that
>     it was out of date and that Anthony's FreeSWITCH::CLient would be a
>     better choice.  I have been able to try out the perl FreeSWITCH Client
>     example on the wiki and  send out some calls using "bgapi"...but I am
>     not sure how to handle the multiple threads.  Would I just deal
>     with all
>     events in a loop similar to what is in the wiki client example?
>
>     Now I am really confused as to how to tackle this.  Not an expert with
>     any particular language...but it seems that scripting may be nice
>     from a
>     maintenance point of view??
>
>     Any further thoughts or examples to get me on my way?  I am really
>     open
>     as to the approach to take.
>
>     Thanks
>
>     So now I don't know what direction to go to get this started.
>     Jan Berger wrote:
>     > You don't need to pull a db these day's, you can use triggers to
>     signal your
>     > application - a bit depending on what db you use and what
>     scripting they
>     > offer beyond SQL.
>     >
>     > -----Original Message-----
>     > From: freeswitch-users-bounces at lists.freeswitch.org
>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>
>     > [mailto:freeswitch-users-bounces at lists.freeswitch.org
>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf
>     Of Phone
>     > Sent: 13. mai 2010 01:46
>     > To: freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>
>     > Subject: Re: [Freeswitch-users] Questions on Building an
>     application for
>     > FreeSWITCH
>     >
>     > Thanks for sharing your experience.
>     >
>     > We would like to get over to linux with this.
>     >
>     > I am still trying to get my head around how this generally works.  I
>     > guess we would write a client to sit and run on the linux box
>     where it
>     > could periodically check the DB and then talk sockets to FS to
>     make the
>     > calls?
>     >
>     > Can you point me to examples of talking sockets?
>     >
>     >
>     >
>     >
>     >
>     > Kristian Kielhofner wrote:
>     >
>     >> Glad to hear.
>     >>
>     >> I still say to build the app and logic completely separately
>     and connect
>     >>
>     > it
>     >
>     >> to FreeSWITCH over the socket using ESL.
>     >>
>     >>
>     >> --
>     >> Kristian Kielhofner
>     >> http://blog.krisk.org
>     >>
>     >> ----- Original Message -----
>     >> From: freeswitch-users-bounces at lists.freeswitch.org
>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>
>     >> <freeswitch-users-bounces at lists.freeswitch.org
>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>>
>     >> To: freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>
>     >> <freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>>
>     >> Sent: Wed May 12 18:28:13 2010
>     >> Subject: Re: [Freeswitch-users] Questions on Building an
>     application
>     >> forFreeSWITCHThis is legit.  We are not into harassing anyone.
>      There
>     >> are too many
>     >> people doing that already.
>     >>
>     >> We currently have this working on a Windows/Dialogic platform,
>     but would
>     >> like to get it over to FreeSWITCH for a number of reasons.  It
>     is always
>     >> nice to avoid spending alot of time developing using a method
>     that you
>     >> later find has serious issues or limitations and then having to
>     change
>     >> horses and start over.  I just thought I would tap on someone
>     else's
>     >> wisdom in regards to a good way to talk to FS that works well.
>      I did
>     >> notice that the wiki talks like LUA is the preferred way to go.
>     >>
>     >> Thanks
>     >>
>     >>
>     >>
>     >> Michael Collins wrote:
>     >>
>     >>
>     >>> On Wed, May 12, 2010 at 2:50 PM, Kristian Kielhofner
>     >>> <kris at kriskinc.com <mailto:kris at kriskinc.com>
>     <mailto:kris at kriskinc.com <mailto:kris at kriskinc.com>>> wrote:
>     >>>
>     >>>     Are you going to be calling my cell phone to ask if I owe
>     $10,000
>     >>>     or more to
>     >>>     the IRS?
>     >>>
>     >>>     Seriously, the world does not need another robodialer to
>     harass
>     >>>     people...
>     >>>
>     >>> Agreed, however there are legitimate use cases for this, like a
>     >>> retailer calling to say an order is in, or reminder calls sent
>     out by
>     >>> a doctor's office. But yes, if it's dialing for dollars then
>     I'm with
>     >>> you: Por favor! No mas!
>     >>> -MC
>     >>>
>     >>>
>     >>>
>     >>>     With that being said, write your app in whatever you want
>     using
>     >>>     ESL and
>     >>>     connect to the socket to originate calls and listen for
>     events.
>     >>>
>     >>>
>     >>>     --
>     >>>     Kristian Kielhofner
>     >>>     http://blog.krisk.org
>     >>>
>     >>>     ----- Original Message -----
>     >>>     From: freeswitch-users-bounces at lists.freeswitch.org
>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>
>     >>>     <mailto:freeswitch-users-bounces at lists.freeswitch.org
>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>>
>     >>>     <freeswitch-users-bounces at lists.freeswitch.org
>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>
>     >>>     <mailto:freeswitch-users-bounces at lists.freeswitch.org
>     <mailto:freeswitch-users-bounces at lists.freeswitch.org>>>
>     >>>     To: freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>
>     >>>     <mailto:freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>>
>     >>>     <freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>
>     >>>     <mailto:freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>>>
>     >>>     Sent: Wed May 12 15:40:58 2010
>     >>>     Subject: [Freeswitch-users] Questions on Building an
>     application
>     >>>     forFreeSWITCHI am looking for a boot in the right
>     direction on the
>     >>>     following project.
>     >>>
>     >>>     I am looking to build an "application" with FS where it will
>     >>>     frequently
>     >>>     check a database and then make an outbound call when a
>     record with
>     >>>     certain conditions is found in the DB.  It needs to
>     determine if the
>     >>>     call was then answered by a machine/vm, busy, or answered by a
>     >>>
>     > person,
>     >
>     >>>     play a message and get a response back to be recorded in
>     the db.
>     >>>      It may
>     >>>     find many calls that need to be made at the same time.  It
>     will be
>     >>>     running on a PRI.
>     >>>
>     >>>     I am confused as to what approach to use to accomplish this.
>     >>>
>     >>>     Lua Script, Perl Script, Javascript, Phython, PHP,
>     Ruby....etc.
>     >>>     Liverpie, ESL....etc.
>     >>>     Or a totally different approach.
>     >>>
>     >>>     I realize that there is not only one "correct" way to
>     build this,
>     >>>     but I
>     >>>     am looking for advise from those who have done this type
>     of thing
>     >>>     before.  I value the opinion of those who have gone down
>     this road
>     >>>
>     > and
>     >
>     >>>     would be willing to share their thoughts on a recommended
>     path to
>     >>>
>     > take
>     >
>     >>>     to accomplish this.
>     >>>
>     >>>     Thanks
>     >>>
>     >>>
>     >>>     _______________________________________________
>     >>>     FreeSWITCH-users mailing list
>     >>>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >>>     <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>     >>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >>>
>     >>>
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >>>     http://www.freeswitch.org
>     >>>
>     >>>     _______________________________________________
>     >>>     FreeSWITCH-users mailing list
>     >>>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >>>     <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>     >>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >>>
>     >>>
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >>>     http://www.freeswitch.org
>     >>>
>     >>>
>     >>>
>
------------------------------------------------------------------------
>     >>>
>     >>> _______________________________________________
>     >>> FreeSWITCH-users mailing list
>     >>> FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >>>
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >>> http://www.freeswitch.org
>     >>>
>     >>>
>     >>>
>     >> _______________________________________________
>     >> FreeSWITCH-users mailing list
>     >> FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >>
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >> http://www.freeswitch.org
>     >>
>     >> _______________________________________________
>     >> FreeSWITCH-users mailing list
>     >> FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >>
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >> http://www.freeswitch.org
>     >>
>     >>
>     >
>     >
>     > _______________________________________________
>     > FreeSWITCH-users mailing list
>     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > http://www.freeswitch.org
>     >
>     >
>     > _______________________________________________
>     > FreeSWITCH-users mailing list
>     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > http://www.freeswitch.org
>     >
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


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



From msc at freeswitch.org  Tue May 25 11:45:50 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 25 May 2010 11:45:50 -0700
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
In-Reply-To: <9250A9FF05DB4B879DEF6ECE8E3FFAC1@dell9400>
References: <490aedfdd7c8ac680bbe0d7f68c786de@mail.gmail.com>
	<4BEB3DD4.9070807@gmail.com>
	<EC392AB972A94F8E8D8369C802771451@dell9400>
	<4BFBE6C2.5000703@gmail.com>
	<AANLkTilsHn9znajzF0Jj-QDu4je9QW7CeymuBYE0SyKa@mail.gmail.com>
	<4BFBEF7C.9020107@gmail.com>
	<9250A9FF05DB4B879DEF6ECE8E3FFAC1@dell9400>
Message-ID: <AANLkTilU2OwgH0Qk_Qoc3R7ErPO_Z_VRhePusetkRkeW@mail.gmail.com>

On Tue, May 25, 2010 at 11:38 AM, Jan Berger <jan.berger at video24.no> wrote:

> Actually - before you get "to smart" - may I suggest that you start writing
> - or improving - the getting started sections of the doc. Address the areas
> where you struggle and let others benefit from your work.
>
> I have been through similar issues myself - FS is one of the easier
> projects
> to work with once you get under the hood, but you basically need to evolve
> to the level where you read the source code.
>
> And if you can wait 2+ months for "the book" then that should help as well.
:D
-MC
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From msc at freeswitch.org  Tue May 25 11:46:45 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 25 May 2010 11:46:45 -0700
Subject: [Freeswitch-users] IRC log for #openzap
In-Reply-To: <AANLkTimnxUUYtYdrk7_3T9_GNFn0BusFxkXO4cdZu5yb@mail.gmail.com>
References: <AANLkTimnxUUYtYdrk7_3T9_GNFn0BusFxkXO4cdZu5yb@mail.gmail.com>
Message-ID: <AANLkTimGdAvt0PToxgLAhxqOL4kpCeZSkGzpsFlQPcaf@mail.gmail.com>

what do you need? I don't believe this file is on a public server but if
there's a specific conversation or date that you need we can probably fish
it out.
-MC

On Mon, May 24, 2010 at 11:38 PM, Nagalenoj H. <nagalenoj at gmail.com> wrote:

> Hi friends,
>      Where is the openzap channel's IRC log available? Has someone tried
> downloading it already?
>
> --
> Regards,
> Nagalenoj H.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Tue May 25 12:29:08 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 25 May 2010 15:29:08 -0400
Subject: [Freeswitch-users] get a freeswitch var from PHP
References: <A9B2CF8578BA4764A5313248EDCE13BE@MOBILEE1705><E26DC27E-D1C7-483F-98BA-325A679E2572@gmail.com><16132F5DC2A44121B3EF6A6C8A3B55F8@MOBILEE1705><3D453DAA-610E-43DF-BADD-C81AB5861C8C@freeswitch.org>
	<AANLkTilFIguFmRDUXwWCqIffCOfqTyDHdm294_3r0x7T@mail.gmail.com>
Message-ID: <5BA612D437704CDA8BFD6DB2C0AABCB7@MOBILEE1705>

well explained...

thanks Mike !
  ----- Original Message ----- 
  From: Michael Collins 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, May 25, 2010 2:23 PM
  Subject: Re: [Freeswitch-users] get a freeswitch var from PHP





  On Tue, May 25, 2010 at 10:42 AM, Brian West <brian at freeswitch.org> wrote:

    fs_cli -x



  So if you know the uuid of the call this do this:

  fs_cli -x "uuid_getvar <uuid> <varname>"


  Here's an example. From fs_cli I did this:
  originate {my_test_var=abc123xyz}loopback/9999 9999

  That creates two channels, one of which has 'my_test_var' set to 'abc123xyz'.
  show channels gives me the uuid: dc773eec-4dfe-4e8d-841d-ccfb4848658a 

  At the Linux command line:

  root at freeswitch1 [Tue May 25 11:21 AM]  /usr/local/freeswitch
  <25>:fs_cli -x "uuid_getvar dc773eec-4dfe-4e8d-841d-ccfb4848658a my_test_var"
  abc123xyz

  From ESL you have the $con object where you can do a uuid_getvar as well:
  $var_contents = $con->api('uuid_getvar', "$uuid $var_name");  # Perl example

  Have fun.
  -MC





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


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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From infos at madovsky.org  Tue May 25 12:31:12 2010
From: infos at madovsky.org (Madovsky)
Date: Tue, 25 May 2010 15:31:12 -0400
Subject: [Freeswitch-users] vars set in external trunk
References: <7DE4725D17DE4359B6B6A37D48B1DBB2@MOBILEE1705>
	<5567677B-14AF-49B5-B889-0E720FB3F159@freeswitch.org>
Message-ID: <0C21929CB49E49D7BBBDBDFE6E330B9E@MOBILEE1705>

Oops. thanks
  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Tuesday, May 25, 2010 2:00 PM
  Subject: Re: [Freeswitch-users] vars set in external trunk


  The FAQ covers this.


  http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ#Q:_What_is_the_difference_between_using_a_.24.7Bvar.7D_and_.24.24.7Bvar.7D_in_the_configuration_files.3F


  /b


  On May 25, 2010, at 12:57 PM, Madovsky wrote:


    Hi all,

    should I us $${var} or ${var} to call a var from a dialplan set in a external sip profile ?

    Thanks




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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From ranjtech at gmail.com  Tue May 25 12:48:26 2010
From: ranjtech at gmail.com (RR)
Date: Tue, 25 May 2010 15:48:26 -0400
Subject: [Freeswitch-users] Direct inward dialling
In-Reply-To: <491C3F16.5010500@gmx.net>
References: <491B305B.4060307@gmx.net>
	<191c3a030811121335g60ef3a41v41070a8cc3f647de@mail.gmail.com>
	<491BF6DB.5030406@gmx.net>
	<191c3a030811130621p72a4be5cje130874670a7c714@mail.gmail.com>
	<491C3F16.5010500@gmx.net>
Message-ID: <AANLkTin9KcsOPtJybqW9tNatRvRUR_2ND7XnecayCstu@mail.gmail.com>

Hello I want to follow up on this example from YEARS ago. I had tried using
the variable "destination_number" but that didn't work, and I figured that
it was because the To: header doesn't have the destination_number but has
just the URI, so I thought I'd use sip_to_user instead.

We have calls coming in with the following info in the INVITE

From: "16469NNNNNN" <sip:16469NNNNNN;phone-context=+1 at 67.1x.xxx.xxx
:5060;user=phone>;tag=SDru6fc01-gK0c10a887.
To: <sip:011390NNNNNNNNNN;phone-context=+39 at xxx.xxx.xxx.xxx
:5060;user=phone>.
(N and x are obviously being masked for privacy)

I use this info in the dialplan like so

<include>
  <extension name="DIDtest">
    <condition field="ani" expression="^(\+?|\+1?|1?)(6469NNNNNN).*$"
break="never">
        <action application="set" data="effective_caller_id_number=$2"/>
        <action application="set" data="effective_caller_id_name=$2"/>
    </condition>
    <condition field="${sip_to_user}"
expression="^(\+?|\+1?|1?|011?)(390NNNNNNNNNN).$" break="never">
        <action application="set" data="continue_on_fail=false"/>
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set" data="domain_name=$${domain}"/>
        <action application="set" data="bypass_media=true"/>
        <action application="bridge"
data="{sip_invite_domain=${sip_from_host}}sofia/gateway/blade2/$2"/>
    </condition>
  </extension>
</include>

However, the calls aren't passing the condition in this dialplan and thus
not being forwarded to "blade2" server. In fact, even the 011 is not being
stripped off.

What am I doing wrong?

Thanks
\RR


On Thu, Nov 13, 2008 at 10:52 AM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> Thanks,
>
> this works
>
> Best regards
> Peter
>
> Anthony Minessale schrieb:
> > here's another hint.
> >
> > use field="${sip_to_user}"
> >
> >
> > On Thu, Nov 13, 2008 at 3:43 AM, Peter P GMX <Prometheus001 at gmx.net
> > <mailto:Prometheus001 at gmx.net>> wrote:
> >
> >     I checked your hint. The variable sip_to_user has the right value
> >     - but
> >     it is evaluated as empty at the time it is parsed.
> >
> >     Info gives me:
> >     .
> >     .
> >     variable_sip_to_user: [xxxxxxxxx8910]
> >     variable_sip_to_uri: [xxxxxxxxx8910 at my.ip.addre.ss]
> >
> >
> >     Example1,:
> >     I check with condition field="destination_number"
> >     Match Result
> >     2008-11-13 10:24:55 [DEBUG] mod_dialplan_xml.c:117 parse_exten()
> >     Regex:
> >     [Inbound QSC trunk] destination_number(xxxxxxxxx89) =~
> >     /^(xxxxxxxxx89)$/
> >     This works but gives only the DID of the trunk (10 is missing at
> >     the end).
> >
> >     Example2:
> >     I check with condition field="sip_to_user"
> >     Match Result
> >     2008-11-13 10:30:42 [DEBUG] mod_dialplan_xml.c:117 parse_exten()
> >     Regex:
> >     [Inbound QSC trunk] sip_to_user() =~ /^(xxxxxxxxx89[0-9.])$/
> >     2008-11-13 10:30:42 [DEBUG] mod_dialplan_xml.c:119 parse_exten()
> Regex
> >     mismatch
> >     sip_to_user() is empty
> >     Example3:
> >     I check with condition field="variable_sip_to_user"
> >     Match Result
> >     2008-11-13 10:39:53 [DEBUG] mod_dialplan_xml.c:117 parse_exten()
> >     Regex:
> >     [Inbound QSC trunk] variable_sip_to_user() =~ /^(xxxxxxxxx89[0-9.])$/
> >     2008-11-13 10:39:53 [DEBUG] mod_dialplan_xml.c:119 parse_exten()
> Regex
> >     mismatch
> >     variable_sip_to_user() is empty
> >
> >     So at the time of evaluation "sip_to_user" resp.
> >     "variable_sip_to_user"
> >     is empty.
> >
> >     What can I do?
> >
> >     Best regards
> >     Peter
> >
> >
> >     Anthony Minessale schrieb:
> >     > route the call to the "info" app and look for the one that has it.
> >     >
> >     >
> >     > On Wed, Nov 12, 2008 at 1:36 PM, Peter P GMX
> >     <Prometheus001 at gmx.net <mailto:Prometheus001 at gmx.net>
> >     > <mailto:Prometheus001 at gmx.net <mailto:Prometheus001 at gmx.net>>>
> >     wrote:
> >     >
> >     >     Dear all,
> >     >
> >     >     when I get an incoming call from my SIP provider I do receive
> an
> >     >     invite
> >     >     on the trunk number, e.g. in Germany 0xxxxx9. However I have an
> >     >     extension block of 0xxxxx90 to 0xxxxx999.
> >     >     In the dialplan I checked the condition
> >     field="destination_number" but
> >     >     this compares to the number of the trunk 0xxxxx9, so I do
> >     not get the
> >     >     direct inward dialling extensions.
> >     >
> >     >     The SIP to-header contains the direct inward dialling number
> >     >     0xxxxx90 to
> >     >     0xxxxx999 but I have no success in accessing it in the
> dialplan.
> >     >
> >     >     I tried sip_to_user, sip_to_uri, sip_req_user, sip_req_uri,
> >     but no
> >     >     success. They are all empty.
> >     >
> >     >     Any idea which variable to parse?
> >     >
> >     >     Best regards
> >     >     Peter
> >     >
> >     >
> >     >
> >     >
> >     >
> >     >
> >     >     _______________________________________________
> >     >     Freeswitch-users mailing list
> >     >     Freeswitch-users at lists.freeswitch.org
> >     <mailto:Freeswitch-users at lists.freeswitch.org>
> >     >     <mailto:Freeswitch-users at lists.freeswitch.org
> >     <mailto:Freeswitch-users at lists.freeswitch.org>>
> >     >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     >
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     >     http://www.freeswitch.org
> >     >
> >     >
> >     >
> >     >
> >     > --
> >     > Anthony Minessale II
> >     >
> >     > FreeSWITCH http://www.freeswitch.org/
> >     > ClueCon http://www.cluecon.com/
> >     >
> >     > AIM: anthm
> >     > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >     <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> >     > <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >     <mailto:MSN%253Aanthony_minessale at hotmail.com<MSN%25253Aanthony_minessale at hotmail.com>
> >>
> >     > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >     <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> >     > <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >     <mailto:PAYPAL%253Aanthony.minessale at gmail.com<PAYPAL%25253Aanthony.minessale at gmail.com>
> >>
> >     > IRC: irc.freenode.net <http://irc.freenode.net>
> >     <http://irc.freenode.net> #freeswitch
> >     >
> >     > FreeSWITCH Developer Conference
> >     > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >     <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> >     > <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >     <mailto:sip%253A888 at conference.freeswitch.org<sip%25253A888 at conference.freeswitch.org>
> >>
> >     > iax:guest at conference.freeswitch.org/888
> >     <http://iax:guest at conference.freeswitch.org/888>
> >     > <http://iax:guest at conference.freeswitch.org/888>
> >     > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >     <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> >     > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >     <mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org<googletalk%25253Aconf%25252B888 at conference.freeswitch.org>
> >>
> >     > pstn:213-799-1400
> >     >
> >
> ------------------------------------------------------------------------
> >     >
> >     > _______________________________________________
> >     > Freeswitch-users mailing list
> >     > Freeswitch-users at lists.freeswitch.org
> >     <mailto:Freeswitch-users at lists.freeswitch.org>
> >     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     >
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     > http://www.freeswitch.org
> >     >
> >
> >     _______________________________________________
> >     Freeswitch-users mailing list
> >     Freeswitch-users at lists.freeswitch.org
> >     <mailto:Freeswitch-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     http://www.freeswitch.org
> >
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> > IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> > iax:guest at conference.freeswitch.org/888
> > <http://iax:guest at conference.freeswitch.org/888>
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> > pstn:213-799-1400
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From codecomplete at free.fr  Tue May 25 12:49:56 2010
From: codecomplete at free.fr (GillesToo)
Date: Tue, 25 May 2010 12:49:56 -0700 (PDT)
Subject: [Freeswitch-users] make current -> "Failed to add directory
	'libs/openzap'"
Message-ID: <1274816996591-5100197.post@n2.nabble.com>


Hello

After updating a test CentOS 5.4 host with "yum update", I wanted to also
update Freeswitch. I followed the Installation Guide
(http://wiki.freeswitch.org/wiki/Installation_Guide) thusly:

---------
# cd /usr/src/freeswitch

# diff build/modules.conf.in modules.conf
[nothing shown]

# make current
[...]
U    libs/libsndfile/examples/sndfile-play-beos.cpp
U    libs/libsndfile/reconfigure.mk
U   libs/libsndfile
svn: Failed to add directory 'libs/openzap': object of the same name already
exists
make: *** [current] Error 1
---------

Then, I ran "make current" a second time:
---------
# make update
cd . && /bin/sh /usr/src/freeswitch/build/config/missing --run aclocal-1.9 
 cd . && /bin/sh /usr/src/freeswitch/build/config/missing --run automake-1.9
--gnu 
configure.in:898: required file `build/freeswitch.pc.in' not found
Makefile.am:256: ADD_LIBEDIT does not appear in AM_CONDITIONAL
Makefile.am:263: ADD_LIBEDIT does not appear in AM_CONDITIONAL
Makefile.am:268: ADD_ODBC does not appear in AM_CONDITIONAL
make: *** [Makefile.in] Error 1
---------

FWIW, this host has an one-FXO OpenVox PCI card and things were working OK.

Is there something I can try besides wiping out /usr/src/freeswitch/ and
just compiling from scratch?

Thank you.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/make-current-Failed-to-add-directory-libs-openzap-tp5100197p5100197.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From david.ponzone at gmail.com  Tue May 25 13:02:21 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Tue, 25 May 2010 22:02:21 +0200
Subject: [Freeswitch-users] Direct inward dialling
In-Reply-To: <AANLkTin9KcsOPtJybqW9tNatRvRUR_2ND7XnecayCstu@mail.gmail.com>
References: <491B305B.4060307@gmx.net>
	<191c3a030811121335g60ef3a41v41070a8cc3f647de@mail.gmail.com>
	<491BF6DB.5030406@gmx.net>
	<191c3a030811130621p72a4be5cje130874670a7c714@mail.gmail.com>
	<491C3F16.5010500@gmx.net>
	<AANLkTin9KcsOPtJybqW9tNatRvRUR_2ND7XnecayCstu@mail.gmail.com>
Message-ID: <94E6BA0F-9006-4EBB-8609-6B6FEB561AC3@gmail.com>

You mean the SIP TO you receive does not contain a @domain part ?

Hmmm, I am sure some guys around will have some comments to do about  
that :)

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 25/05/2010 ? 21:48, RR a ?crit :

> Hello I want to follow up on this example from YEARS ago. I had  
> tried using the variable "destination_number" but that didn't work,  
> and I figured that it was because the To: header doesn't have the  
> destination_number but has just the URI, so I thought I'd use  
> sip_to_user instead.
>
> We have calls coming in with the following info in the INVITE
>
> From: "16469NNNNNN" <sip:16469NNNNNN;phone-context=+1 at 67.1x.xxx.xxx:5060;user=phone 
> >;tag=SDru6fc01-gK0c10a887.
> To: <sip:011390NNNNNNNNNN;phone-context=+39 at xxx.xxx.xxx.xxx:5060;user=phone 
> >.
> (N and x are obviously being masked for privacy)
>
> I use this info in the dialplan like so
>
> <include>
>   <extension name="DIDtest">
>     <condition field="ani" expression="^(\+?|\+1?|1?)(6469NNNNNN).* 
> $" break="never">
>         <action application="set" data="effective_caller_id_number= 
> $2"/>
>         <action application="set" data="effective_caller_id_name=$2"/>
>     </condition>
>     <condition field="${sip_to_user}" expression="^(\+?|\+1?|1?|011?) 
> (390NNNNNNNNNN).$" break="never">
>         <action application="set" data="continue_on_fail=false"/>
>         <action application="set" data="hangup_after_bridge=true"/>
>         <action application="set" data="domain_name=$${domain}"/>
>         <action application="set" data="bypass_media=true"/>
>         <action application="bridge" data="{sip_invite_domain=$ 
> {sip_from_host}}sofia/gateway/blade2/$2"/>
>     </condition>
>   </extension>
> </include>
>
> However, the calls aren't passing the condition in this dialplan and  
> thus not being forwarded to "blade2" server. In fact, even the 011  
> is not being stripped off.
>
> What am I doing wrong?
>
> Thanks
> \RR
>
>
> On Thu, Nov 13, 2008 at 10:52 AM, Peter P GMX  
> <Prometheus001 at gmx.net> wrote:
> Thanks,
>
> this works
>
> Best regards
> Peter
>
> Anthony Minessale schrieb:
> > here's another hint.
> >
> > use field="${sip_to_user}"
> >
> >
> > On Thu, Nov 13, 2008 at 3:43 AM, Peter P GMX <Prometheus001 at gmx.net
> > <mailto:Prometheus001 at gmx.net>> wrote:
> >
> >     I checked your hint. The variable sip_to_user has the right  
> value
> >     - but
> >     it is evaluated as empty at the time it is parsed.
> >
> >     Info gives me:
> >     .
> >     .
> >     variable_sip_to_user: [xxxxxxxxx8910]
> >     variable_sip_to_uri: [xxxxxxxxx8910 at my.ip.addre.ss]
> >
> >
> >     Example1,:
> >     I check with condition field="destination_number"
> >     Match Result
> >     2008-11-13 10:24:55 [DEBUG] mod_dialplan_xml.c:117 parse_exten()
> >     Regex:
> >     [Inbound QSC trunk] destination_number(xxxxxxxxx89) =~
> >     /^(xxxxxxxxx89)$/
> >     This works but gives only the DID of the trunk (10 is missing at
> >     the end).
> >
> >     Example2:
> >     I check with condition field="sip_to_user"
> >     Match Result
> >     2008-11-13 10:30:42 [DEBUG] mod_dialplan_xml.c:117 parse_exten()
> >     Regex:
> >     [Inbound QSC trunk] sip_to_user() =~ /^(xxxxxxxxx89[0-9.])$/
> >     2008-11-13 10:30:42 [DEBUG] mod_dialplan_xml.c:119  
> parse_exten() Regex
> >     mismatch
> >     sip_to_user() is empty
> >     Example3:
> >     I check with condition field="variable_sip_to_user"
> >     Match Result
> >     2008-11-13 10:39:53 [DEBUG] mod_dialplan_xml.c:117 parse_exten()
> >     Regex:
> >     [Inbound QSC trunk] variable_sip_to_user() =~ / 
> ^(xxxxxxxxx89[0-9.])$/
> >     2008-11-13 10:39:53 [DEBUG] mod_dialplan_xml.c:119  
> parse_exten() Regex
> >     mismatch
> >     variable_sip_to_user() is empty
> >
> >     So at the time of evaluation "sip_to_user" resp.
> >     "variable_sip_to_user"
> >     is empty.
> >
> >     What can I do?
> >
> >     Best regards
> >     Peter
> >
> >
> >     Anthony Minessale schrieb:
> >     > route the call to the "info" app and look for the one that  
> has it.
> >     >
> >     >
> >     > On Wed, Nov 12, 2008 at 1:36 PM, Peter P GMX
> >     <Prometheus001 at gmx.net <mailto:Prometheus001 at gmx.net>
> >     > <mailto:Prometheus001 at gmx.net <mailto:Prometheus001 at gmx.net>>>
> >     wrote:
> >     >
> >     >     Dear all,
> >     >
> >     >     when I get an incoming call from my SIP provider I do  
> receive an
> >     >     invite
> >     >     on the trunk number, e.g. in Germany 0xxxxx9. However I  
> have an
> >     >     extension block of 0xxxxx90 to 0xxxxx999.
> >     >     In the dialplan I checked the condition
> >     field="destination_number" but
> >     >     this compares to the number of the trunk 0xxxxx9, so I do
> >     not get the
> >     >     direct inward dialling extensions.
> >     >
> >     >     The SIP to-header contains the direct inward dialling  
> number
> >     >     0xxxxx90 to
> >     >     0xxxxx999 but I have no success in accessing it in the  
> dialplan.
> >     >
> >     >     I tried sip_to_user, sip_to_uri, sip_req_user,  
> sip_req_uri,
> >     but no
> >     >     success. They are all empty.
> >     >
> >     >     Any idea which variable to parse?
> >     >
> >     >     Best regards
> >     >     Peter
> >     >
> >     >
> >     >
> >     >
> >     >
> >     >
> >     >     _______________________________________________
> >     >     Freeswitch-users mailing list
> >     >     Freeswitch-users at lists.freeswitch.org
> >     <mailto:Freeswitch-users at lists.freeswitch.org>
> >     >     <mailto:Freeswitch-users at lists.freeswitch.org
> >     <mailto:Freeswitch-users at lists.freeswitch.org>>
> >     >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     >
> >     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     >     http://www.freeswitch.org
> >     >
> >     >
> >     >
> >     >
> >     > --
> >     > Anthony Minessale II
> >     >
> >     > FreeSWITCH http://www.freeswitch.org/
> >     > ClueCon http://www.cluecon.com/
> >     >
> >     > AIM: anthm
> >     > MSN:anthony_minessale at hotmail.com
> >     <mailto:MSN%3Aanthony_minessale at hotmail.com>
> >     > <mailto:MSN%3Aanthony_minessale at hotmail.com
> >     <mailto:MSN%253Aanthony_minessale at hotmail.com>>
> >     > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> >     <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> >     > <mailto:PAYPAL%3Aanthony.minessale at gmail.com
> >     <mailto:PAYPAL%253Aanthony.minessale at gmail.com>>
> >     > IRC: irc.freenode.net <http://irc.freenode.net>
> >     <http://irc.freenode.net> #freeswitch
> >     >
> >     > FreeSWITCH Developer Conference
> >     > sip:888 at conference.freeswitch.org
> >     <mailto:sip%3A888 at conference.freeswitch.org>
> >     > <mailto:sip%3A888 at conference.freeswitch.org
> >     <mailto:sip%253A888 at conference.freeswitch.org>>
> >     > iax:guest at conference.freeswitch.org/888
> >     <http://iax:guest at conference.freeswitch.org/888>
> >     > <http://iax:guest at conference.freeswitch.org/888>
> >     > googletalk:conf+888 at conference.freeswitch.org
> >     <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >     > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org
> >     <mailto:googletalk%253Aconf%252B888 at conference.freeswitch.org>>
> >     > pstn:213-799-1400
> >     >
> >      
> ------------------------------------------------------------------------
> >     >
> >     > _______________________________________________
> >     > Freeswitch-users mailing list
> >     > Freeswitch-users at lists.freeswitch.org
> >     <mailto:Freeswitch-users at lists.freeswitch.org>
> >     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     >
> >     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     > http://www.freeswitch.org
> >     >
> >
> >     _______________________________________________
> >     Freeswitch-users mailing list
> >     Freeswitch-users at lists.freeswitch.org
> >     <mailto:Freeswitch-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     http://www.freeswitch.org
> >
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com
> > <mailto:MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> > <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org
> > <mailto:sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > <http://iax:guest at conference.freeswitch.org/888>
> > googletalk:conf+888 at conference.freeswitch.org
> > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:213-799-1400
> >  
> ------------------------------------------------------------------------
> >
> > _______________________________________________
> > Freeswitch-users mailing list
> > Freeswitch-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From msc at freeswitch.org  Tue May 25 13:12:35 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 25 May 2010 13:12:35 -0700
Subject: [Freeswitch-users] Direct inward dialling
In-Reply-To: <AANLkTin9KcsOPtJybqW9tNatRvRUR_2ND7XnecayCstu@mail.gmail.com>
References: <491B305B.4060307@gmx.net>
	<191c3a030811121335g60ef3a41v41070a8cc3f647de@mail.gmail.com>
	<491BF6DB.5030406@gmx.net>
	<191c3a030811130621p72a4be5cje130874670a7c714@mail.gmail.com>
	<491C3F16.5010500@gmx.net>
	<AANLkTin9KcsOPtJybqW9tNatRvRUR_2ND7XnecayCstu@mail.gmail.com>
Message-ID: <AANLkTinPOe6hM0_ZToxxnThCecXCtGyNxtolUkS9w4of@mail.gmail.com>

On Tue, May 25, 2010 at 12:48 PM, RR <ranjtech at gmail.com> wrote:

> Hello I want to follow up on this example from YEARS ago. I had tried using
> the variable "destination_number" but that didn't work, and I figured that
> it was because the To: header doesn't have the destination_number but has
> just the URI, so I thought I'd use sip_to_user instead.
>
> We have calls coming in with the following info in the INVITE
>
> From: "16469NNNNNN" <sip:16469NNNNNN;phone-context=+1 at 67.1x.xxx.xxx
> :5060;user=phone>;tag=SDru6fc01-gK0c10a887.
> To: <sip:011390NNNNNNNNNN;phone-context=+39 at xxx.xxx.xxx.xxx
> :5060;user=phone>.
> (N and x are obviously being masked for privacy)
>
> I use this info in the dialplan like so
>
> <include>
>   <extension name="DIDtest">
>     <condition field="ani" expression="^(\+?|\+1?|1?)(6469NNNNNN).*$"
> break="never">
>         <action application="set" data="effective_caller_id_number=$2"/>
>         <action application="set" data="effective_caller_id_name=$2"/>
>     </condition>
>     <condition field="${sip_to_user}"
> expression="^(\+?|\+1?|1?|011?)(390NNNNNNNNNN).$" break="never">
>         <action application="set" data="continue_on_fail=false"/>
>         <action application="set" data="hangup_after_bridge=true"/>
>         <action application="set" data="domain_name=$${domain}"/>
>         <action application="set" data="bypass_media=true"/>
>         <action application="bridge"
> data="{sip_invite_domain=${sip_from_host}}sofia/gateway/blade2/$2"/>
>     </condition>
>   </extension>
> </include>
>
> However, the calls aren't passing the condition in this dialplan and thus
> not being forwarded to "blade2" server. In fact, even the 011 is not being
> stripped off.
>
> What am I doing wrong?
>

Create a quick test extension that only does an info dump. (See 9992 in
default.xml for an example.) Make a call, look at the info dump, and make
sure that what you think you are getting is really what you are getting. :)

-MC
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From ranjtech at gmail.com  Tue May 25 13:28:53 2010
From: ranjtech at gmail.com (RR)
Date: Tue, 25 May 2010 16:28:53 -0400
Subject: [Freeswitch-users] Direct inward dialling
In-Reply-To: <AANLkTinPOe6hM0_ZToxxnThCecXCtGyNxtolUkS9w4of@mail.gmail.com>
References: <491B305B.4060307@gmx.net>
	<191c3a030811121335g60ef3a41v41070a8cc3f647de@mail.gmail.com>
	<491BF6DB.5030406@gmx.net>
	<191c3a030811130621p72a4be5cje130874670a7c714@mail.gmail.com>
	<491C3F16.5010500@gmx.net>
	<AANLkTin9KcsOPtJybqW9tNatRvRUR_2ND7XnecayCstu@mail.gmail.com>
	<AANLkTinPOe6hM0_ZToxxnThCecXCtGyNxtolUkS9w4of@mail.gmail.com>
Message-ID: <AANLkTimYlmCTFspJo0531NVf5W6QOpJNs_fzFtB8Vi3e@mail.gmail.com>

Hi Guys,

Thanks for the quick feedback

David, no we're getting the full URI with the domain part intact, just
nothing before the "<" braces

Michael, I already tried the info app and we get

variable_sip_to_user: [011390NNNNNNNNNN;phone-context=+39]
variable_sip_to_uri: [011390NNNNNNNNNN;phone-context=+39 at 208.xx.xxx.xxx
:5060]

Thanks
RR


On Tue, May 25, 2010 at 4:12 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Tue, May 25, 2010 at 12:48 PM, RR <ranjtech at gmail.com> wrote:
>
>> Hello I want to follow up on this example from YEARS ago. I had tried
>> using the variable "destination_number" but that didn't work, and I figured
>> that it was because the To: header doesn't have the destination_number but
>> has just the URI, so I thought I'd use sip_to_user instead.
>>
>> We have calls coming in with the following info in the INVITE
>>
>> From: "16469NNNNNN" <sip:16469NNNNNN;phone-context=+1 at 67.1x.xxx.xxx
>> :5060;user=phone>;tag=SDru6fc01-gK0c10a887.
>> To: <sip:011390NNNNNNNNNN;phone-context=+39 at xxx.xxx.xxx.xxx
>> :5060;user=phone>.
>> (N and x are obviously being masked for privacy)
>>
>> I use this info in the dialplan like so
>>
>> <include>
>>   <extension name="DIDtest">
>>     <condition field="ani" expression="^(\+?|\+1?|1?)(6469NNNNNN).*$"
>> break="never">
>>         <action application="set" data="effective_caller_id_number=$2"/>
>>         <action application="set" data="effective_caller_id_name=$2"/>
>>     </condition>
>>     <condition field="${sip_to_user}"
>> expression="^(\+?|\+1?|1?|011?)(390NNNNNNNNNN).$" break="never">
>>         <action application="set" data="continue_on_fail=false"/>
>>         <action application="set" data="hangup_after_bridge=true"/>
>>         <action application="set" data="domain_name=$${domain}"/>
>>         <action application="set" data="bypass_media=true"/>
>>         <action application="bridge"
>> data="{sip_invite_domain=${sip_from_host}}sofia/gateway/blade2/$2"/>
>>     </condition>
>>   </extension>
>> </include>
>>
>> However, the calls aren't passing the condition in this dialplan and thus
>> not being forwarded to "blade2" server. In fact, even the 011 is not being
>> stripped off.
>>
>> What am I doing wrong?
>>
>
> Create a quick test extension that only does an info dump. (See 9992 in
> default.xml for an example.) Make a call, look at the info dump, and make
> sure that what you think you are getting is really what you are getting. :)
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From testeador01 at gmail.com  Tue May 25 13:40:23 2010
From: testeador01 at gmail.com (Milena)
Date: Tue, 25 May 2010 15:40:23 -0500
Subject: [Freeswitch-users] make current -> "Failed to add directory
	'libs/openzap'"
In-Reply-To: <1274816996591-5100197.post@n2.nabble.com>
References: <1274816996591-5100197.post@n2.nabble.com>
Message-ID: <AANLkTimLE3l8hYCbyNYZPk3qcRpoCtXcVH5c3lunxaJv@mail.gmail.com>

hmmm

I got that error once like 2 months ago... anyways i think the
"how-to-fix-it" is somewhere on the lists and has to do with deleting and
rebuilding, you can look for it, but fixing it is not worth it because if
you want to update you should be UPDATING to GIT!!!

Have a nice day.

2010/5/25 GillesToo <codecomplete at free.fr>

>
> Hello
>
> After updating a test CentOS 5.4 host with "yum update", I wanted to also
> update Freeswitch. I followed the Installation Guide
> (http://wiki.freeswitch.org/wiki/Installation_Guide) thusly:
>
> ---------
> # cd /usr/src/freeswitch
>
> # diff build/modules.conf.in modules.conf
> [nothing shown]
>
> # make current
> [...]
> U    libs/libsndfile/examples/sndfile-play-beos.cpp
> U    libs/libsndfile/reconfigure.mk
> U   libs/libsndfile
> svn: Failed to add directory 'libs/openzap': object of the same name
> already
> exists
> make: *** [current] Error 1
> ---------
>
> Then, I ran "make current" a second time:
> ---------
> # make update
> cd . && /bin/sh /usr/src/freeswitch/build/config/missing --run aclocal-1.9
>  cd . && /bin/sh /usr/src/freeswitch/build/config/missing --run
> automake-1.9
> --gnu
> configure.in:898: required file `build/freeswitch.pc.in' not found
> Makefile.am:256: ADD_LIBEDIT does not appear in AM_CONDITIONAL
> Makefile.am:263: ADD_LIBEDIT does not appear in AM_CONDITIONAL
> Makefile.am:268: ADD_ODBC does not appear in AM_CONDITIONAL
> make: *** [Makefile.in] Error 1
> ---------
>
> FWIW, this host has an one-FXO OpenVox PCI card and things were working OK.
>
> Is there something I can try besides wiping out /usr/src/freeswitch/ and
> just compiling from scratch?
>
> Thank you.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/make-current-Failed-to-add-directory-libs-openzap-tp5100197p5100197.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From david.ponzone at gmail.com  Tue May 25 13:40:58 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Tue, 25 May 2010 22:40:58 +0200
Subject: [Freeswitch-users] Direct inward dialling
In-Reply-To: <AANLkTimYlmCTFspJo0531NVf5W6QOpJNs_fzFtB8Vi3e@mail.gmail.com>
References: <491B305B.4060307@gmx.net>
	<191c3a030811121335g60ef3a41v41070a8cc3f647de@mail.gmail.com>
	<491BF6DB.5030406@gmx.net>
	<191c3a030811130621p72a4be5cje130874670a7c714@mail.gmail.com>
	<491C3F16.5010500@gmx.net>
	<AANLkTin9KcsOPtJybqW9tNatRvRUR_2ND7XnecayCstu@mail.gmail.com>
	<AANLkTinPOe6hM0_ZToxxnThCecXCtGyNxtolUkS9w4of@mail.gmail.com>
	<AANLkTimYlmCTFspJo0531NVf5W6QOpJNs_fzFtB8Vi3e@mail.gmail.com>
Message-ID: <072A175C-9004-4C14-90EC-9A93A8453787@gmail.com>

Which means there is no @ in the sip: part of the SIP To field. Only  
in the phone-context part.
FS uses the @ to split the strings into pieces, and then in your case,  
it fails as one is missing.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 25/05/2010 ? 22:28, RR a ?crit :

> Hi Guys,
>
> Thanks for the quick feedback
>
> David, no we're getting the full URI with the domain part intact,  
> just nothing before the "<" braces
>
> Michael, I already tried the info app and we get
>
> variable_sip_to_user: [011390NNNNNNNNNN;phone-context=+39]
> variable_sip_to_uri: [011390NNNNNNNNNN;phone-context=+39 at 208.xx.xxx.xxx 
> :5060]
>
> Thanks
> RR
>
>
> On Tue, May 25, 2010 at 4:12 PM, Michael Collins  
> <msc at freeswitch.org> wrote:
>
>
> On Tue, May 25, 2010 at 12:48 PM, RR <ranjtech at gmail.com> wrote:
> Hello I want to follow up on this example from YEARS ago. I had  
> tried using the variable "destination_number" but that didn't work,  
> and I figured that it was because the To: header doesn't have the  
> destination_number but has just the URI, so I thought I'd use  
> sip_to_user instead.
>
> We have calls coming in with the following info in the INVITE
>
> From: "16469NNNNNN" <sip:16469NNNNNN;phone-context=+1 at 67.1x.xxx.xxx:5060;user=phone 
> >;tag=SDru6fc01-gK0c10a887.
> To: <sip:011390NNNNNNNNNN;phone-context=+39 at xxx.xxx.xxx.xxx:5060;user=phone 
> >.
> (N and x are obviously being masked for privacy)
>
> I use this info in the dialplan like so
>
> <include>
>   <extension name="DIDtest">
>     <condition field="ani" expression="^(\+?|\+1?|1?)(6469NNNNNN).* 
> $" break="never">
>         <action application="set" data="effective_caller_id_number= 
> $2"/>
>         <action application="set" data="effective_caller_id_name=$2"/>
>     </condition>
>     <condition field="${sip_to_user}" expression="^(\+?|\+1?|1?|011?) 
> (390NNNNNNNNNN).$" break="never">
>         <action application="set" data="continue_on_fail=false"/>
>         <action application="set" data="hangup_after_bridge=true"/>
>         <action application="set" data="domain_name=$${domain}"/>
>         <action application="set" data="bypass_media=true"/>
>         <action application="bridge" data="{sip_invite_domain=$ 
> {sip_from_host}}sofia/gateway/blade2/$2"/>
>     </condition>
>   </extension>
> </include>
>
> However, the calls aren't passing the condition in this dialplan and  
> thus not being forwarded to "blade2" server. In fact, even the 011  
> is not being stripped off.
>
> What am I doing wrong?
>
> Create a quick test extension that only does an info dump. (See 9992  
> in default.xml for an example.) Make a call, look at the info dump,  
> and make sure that what you think you are getting is really what you  
> are getting. :)
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From anthony.minessale at gmail.com  Tue May 25 13:49:45 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 25 May 2010 15:49:45 -0500
Subject: [Freeswitch-users] Direct inward dialling
In-Reply-To: <072A175C-9004-4C14-90EC-9A93A8453787@gmail.com>
References: <491B305B.4060307@gmx.net>
	<191c3a030811121335g60ef3a41v41070a8cc3f647de@mail.gmail.com>
	<491BF6DB.5030406@gmx.net>
	<191c3a030811130621p72a4be5cje130874670a7c714@mail.gmail.com>
	<491C3F16.5010500@gmx.net>
	<AANLkTin9KcsOPtJybqW9tNatRvRUR_2ND7XnecayCstu@mail.gmail.com>
	<AANLkTinPOe6hM0_ZToxxnThCecXCtGyNxtolUkS9w4of@mail.gmail.com>
	<AANLkTimYlmCTFspJo0531NVf5W6QOpJNs_fzFtB8Vi3e@mail.gmail.com>
	<072A175C-9004-4C14-90EC-9A93A8453787@gmail.com>
Message-ID: <AANLkTinxYIG6S30XMB5GUVuWsXlyhG_slqc_e0PQy0wM@mail.gmail.com>

did you turn up the debug (press f8 or type console loglevel debug)
The debug logs will show the data being passed into the regex and the
results.

P.S.
I hope only your example is from years ago and not your copy of FS.


On Tue, May 25, 2010 at 3:40 PM, David Ponzone <david.ponzone at gmail.com>wrote:

> Which means there is no @ in the sip: part of the SIP To field. Only in the
> phone-context part.
> FS uses the @ to split the strings into pieces, and then in your case, it
> fails as one is missing.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 25/05/2010 ? 22:28, RR a ?crit :
>
> Hi Guys,
>
> Thanks for the quick feedback
>
> David, no we're getting the full URI with the domain part intact, just
> nothing before the "<" braces
>
> Michael, I already tried the info app and we get
>
> variable_sip_to_user: [011390NNNNNNNNNN;phone-context=+39]
> variable_sip_to_uri: [011390NNNNNNNNNN;phone-context=+39 at 208.xx.xxx.xxx
> :5060]
>
> Thanks
> RR
>
>
> On Tue, May 25, 2010 at 4:12 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Tue, May 25, 2010 at 12:48 PM, RR <ranjtech at gmail.com> wrote:
>>
>>> Hello I want to follow up on this example from YEARS ago. I had tried
>>> using the variable "destination_number" but that didn't work, and I figured
>>> that it was because the To: header doesn't have the destination_number but
>>> has just the URI, so I thought I'd use sip_to_user instead.
>>>
>>> We have calls coming in with the following info in the INVITE
>>>
>>> From: "16469NNNNNN" <
>>> sip:16469NNNNNN;phone-context=+1 at 67.1x.xxx.xxx:5060;user=phone
>>> >;tag=SDru6fc01-gK0c10a887.
>>> To: <
>>> sip:011390NNNNNNNNNN;phone-context=+39 at xxx.xxx.xxx.xxx:5060;user=phone>.
>>> (N and x are obviously being masked for privacy)
>>>
>>> I use this info in the dialplan like so
>>>
>>> <include>
>>>   <extension name="DIDtest">
>>>     <condition field="ani" expression="^(\+?|\+1?|1?)(6469NNNNNN).*$"
>>> break="never">
>>>         <action application="set" data="effective_caller_id_number=$2"/>
>>>         <action application="set" data="effective_caller_id_name=$2"/>
>>>     </condition>
>>>     <condition field="${sip_to_user}"
>>> expression="^(\+?|\+1?|1?|011?)(390NNNNNNNNNN).$" break="never">
>>>         <action application="set" data="continue_on_fail=false"/>
>>>         <action application="set" data="hangup_after_bridge=true"/>
>>>         <action application="set" data="domain_name=$${domain}"/>
>>>         <action application="set" data="bypass_media=true"/>
>>>         <action application="bridge"
>>> data="{sip_invite_domain=${sip_from_host}}sofia/gateway/blade2/$2"/>
>>>     </condition>
>>>   </extension>
>>> </include>
>>>
>>> However, the calls aren't passing the condition in this dialplan and thus
>>> not being forwarded to "blade2" server. In fact, even the 011 is not being
>>> stripped off.
>>>
>>> What am I doing wrong?
>>>
>>
>> Create a quick test extension that only does an info dump. (See 9992 in
>> default.xml for an example.) Make a call, look at the info dump, and make
>> sure that what you think you are getting is really what you are getting. :)
>>
>> -MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
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
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From codecomplete at free.fr  Tue May 25 14:13:47 2010
From: codecomplete at free.fr (GillesToo)
Date: Tue, 25 May 2010 14:13:47 -0700 (PDT)
Subject: [Freeswitch-users] make current -> "Failed to add directory
	'libs/openzap'"
In-Reply-To: <AANLkTimLE3l8hYCbyNYZPk3qcRpoCtXcVH5c3lunxaJv@mail.gmail.com>
References: <1274816996591-5100197.post@n2.nabble.com>
	<AANLkTimLE3l8hYCbyNYZPk3qcRpoCtXcVH5c3lunxaJv@mail.gmail.com>
Message-ID: <1274822027357-5100501.post@n2.nabble.com>


Thanks for the tip. I'll remove /usr/src/freeswitch/ and start afresh, then.

Since http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start#Install_from_Git
includes this:
-------
cd /usr/src
sudo apt-get install git-core subversion build-essential autoconf automake
libtool libncurses5 libncurses5-dev
git clone git://git.freeswitch.org/freeswitch.git
cd freeswitch
-------

... I gave this a try: "yum install git-core subversion build-essential
autoconf automake libtool libncurses5 libncurses5-dev", but it ended up
with:

-------
Setting up Install Process
No package git-core available.
Package subversion-1.4.2-4.el5_3.1.i386 already installed and latest version
No package build-essential available.
Package autoconf-2.59-12.noarch already installed and latest version
Package automake-1.9.6-2.3.el5.noarch already installed and latest version
Package libtool-1.5.22-7.el5_4.i386 already installed and latest version
No package libncurses5 available.
No package libncurses5-dev available.
Nothing to do
-------

Does someone know what parts I need to install Git on CentOS, so I can
download the latest source code?

Thank you.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/make-current-Failed-to-add-directory-libs-openzap-tp5100197p5100501.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From testeador01 at gmail.com  Tue May 25 14:35:34 2010
From: testeador01 at gmail.com (Milena)
Date: Tue, 25 May 2010 16:35:34 -0500
Subject: [Freeswitch-users] make current -> "Failed to add directory
	'libs/openzap'"
In-Reply-To: <1274822027357-5100501.post@n2.nabble.com>
References: <1274816996591-5100197.post@n2.nabble.com>
	<AANLkTimLE3l8hYCbyNYZPk3qcRpoCtXcVH5c3lunxaJv@mail.gmail.com>
	<1274822027357-5100501.post@n2.nabble.com>
Message-ID: <AANLkTimrQz07P-M1fLLMx5cd9ErVMJFjq_gS0sv9BfKP@mail.gmail.com>

cd /usr/src
yum install gettext-devel expat-devel curl-devel zlib-devel openssl-devel
bzip2
wget http://kernel.org/pub/software/scm/git/git-1.7.1.tar.bz2
tar -xvjf git-1.7.1.tar.bz2
cd git-1.7.1
make prefix=/usr/local all
make prefix=/usr/local install

2010/5/25 GillesToo <codecomplete at free.fr>

>
> Thanks for the tip. I'll remove /usr/src/freeswitch/ and start afresh,
> then.
>
> Since http://wiki.freeswitch.org/wiki/Ubuntu_Quick_Start#Install_from_Git
> includes this:
> -------
> cd /usr/src
> sudo apt-get install git-core subversion build-essential autoconf automake
> libtool libncurses5 libncurses5-dev
> git clone git://git.freeswitch.org/freeswitch.git
> cd freeswitch
> -------
>
> ... I gave this a try: "yum install git-core subversion build-essential
> autoconf automake libtool libncurses5 libncurses5-dev", but it ended up
> with:
>
> -------
> Setting up Install Process
> No package git-core available.
> Package subversion-1.4.2-4.el5_3.1.i386 already installed and latest
> version
> No package build-essential available.
> Package autoconf-2.59-12.noarch already installed and latest version
> Package automake-1.9.6-2.3.el5.noarch already installed and latest version
> Package libtool-1.5.22-7.el5_4.i386 already installed and latest version
> No package libncurses5 available.
> No package libncurses5-dev available.
> Nothing to do
> -------
>
> Does someone know what parts I need to install Git on CentOS, so I can
> download the latest source code?
>
> Thank you.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/make-current-Failed-to-add-directory-libs-openzap-tp5100197p5100501.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From bobc at devassert.com  Tue May 25 14:40:40 2010
From: bobc at devassert.com (Bob Coleman)
Date: Wed, 26 May 2010 09:40:40 +1200
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
In-Reply-To: <AANLkTilU2OwgH0Qk_Qoc3R7ErPO_Z_VRhePusetkRkeW@mail.gmail.com>
References: <490aedfdd7c8ac680bbe0d7f68c786de@mail.gmail.com>
	<4BEB3DD4.9070807@gmail.com>
	<EC392AB972A94F8E8D8369C802771451@dell9400>
	<4BFBE6C2.5000703@gmail.com>
	<AANLkTilsHn9znajzF0Jj-QDu4je9QW7CeymuBYE0SyKa@mail.gmail.com>
	<4BFBEF7C.9020107@gmail.com>
	<9250A9FF05DB4B879DEF6ECE8E3FFAC1@dell9400>
	<AANLkTilU2OwgH0Qk_Qoc3R7ErPO_Z_VRhePusetkRkeW@mail.gmail.com>
Message-ID: <AANLkTikXWvEB_C3jqOWLHaVcns9-U5Hok-5MqgE_uzzZ@mail.gmail.com>

Have just recently completed a project to convert an old windows
dialogic application(vb6) to FreeSWITCH, would reccommend using the
ESL, was able to map the old dialogic calls to the ESL calls pretty
easily. We used a mixture of inbound and outbound sockets, as we have
people dialing us, not just dialing out etc.

With the dialogic you open a port and make the call and handle the
dtmf, with freeswitch you create a socket connection to FreeSWITCH to
dial the number and then hand it off to an extension for processing
the dtmf(that is one approach any way)

Bob

On Wed, May 26, 2010 at 6:45 AM, Michael Collins <msc at freeswitch.org> wrote:
>
>
> On Tue, May 25, 2010 at 11:38 AM, Jan Berger <jan.berger at video24.no> wrote:
>>
>> Actually - before you get "to smart" - may I suggest that you start
>> writing
>> - or improving - the getting started sections of the doc. Address the
>> areas
>> where you struggle and let others benefit from your work.
>>
>> I have been through similar issues myself - FS is one of the easier
>> projects
>> to work with once you get under the hood, but you basically need to evolve
>> to the level where you read the source code.
>>
> And if you can wait 2+ months for "the book" then that should help as well.
> :D
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From ranjtech at gmail.com  Tue May 25 14:43:37 2010
From: ranjtech at gmail.com (RR)
Date: Tue, 25 May 2010 17:43:37 -0400
Subject: [Freeswitch-users] Direct inward dialling
In-Reply-To: <AANLkTinxYIG6S30XMB5GUVuWsXlyhG_slqc_e0PQy0wM@mail.gmail.com>
References: <491B305B.4060307@gmx.net>
	<191c3a030811121335g60ef3a41v41070a8cc3f647de@mail.gmail.com>
	<491BF6DB.5030406@gmx.net>
	<191c3a030811130621p72a4be5cje130874670a7c714@mail.gmail.com>
	<491C3F16.5010500@gmx.net>
	<AANLkTin9KcsOPtJybqW9tNatRvRUR_2ND7XnecayCstu@mail.gmail.com>
	<AANLkTinPOe6hM0_ZToxxnThCecXCtGyNxtolUkS9w4of@mail.gmail.com>
	<AANLkTimYlmCTFspJo0531NVf5W6QOpJNs_fzFtB8Vi3e@mail.gmail.com>
	<072A175C-9004-4C14-90EC-9A93A8453787@gmail.com>
	<AANLkTinxYIG6S30XMB5GUVuWsXlyhG_slqc_e0PQy0wM@mail.gmail.com>
Message-ID: <AANLkTinjUuCyrK6eVSalV6fUVNFM-j_zlvrzRxq3lJZQ@mail.gmail.com>

Hi Anthony,

this is what I see in the debug:

Dialplan: sofia/external/16469NNNNNN Regex (FAIL) [DIDtest]
${sip_to_user}(011390NNNNNNNNNN;phone-context=+39) =~
/^(\+?|\+1?|1?|011?)(390NNNNNNNNNN).$/ break=on-false

and then it moves on to another dialplan xml file.

please note that info app shows:

variable_sip_to_user: [011390NNNNNNNNNN;phone-context=+39]

shouldn't this actually match??

Oh and yes, the copy of FS is pretty old but this is a production system
which gets 24 / 7 traffic so the upgrade is being pushed and pushed :(

you think this is simply because it's an old build?

Thanks
RR


On Tue, May 25, 2010 at 4:49 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> did you turn up the debug (press f8 or type console loglevel debug)
> The debug logs will show the data being passed into the regex and the
> results.
>
> P.S.
> I hope only your example is from years ago and not your copy of FS.
>
>
> On Tue, May 25, 2010 at 3:40 PM, David Ponzone <david.ponzone at gmail.com>wrote:
>
>> Which means there is no @ in the sip: part of the SIP To field. Only in
>> the phone-context part.
>> FS uses the @ to split the strings into pieces, and then in your case, it
>> fails as one is missing.
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 25/05/2010 ? 22:28, RR a ?crit :
>>
>> Hi Guys,
>>
>> Thanks for the quick feedback
>>
>> David, no we're getting the full URI with the domain part intact, just
>> nothing before the "<" braces
>>
>> Michael, I already tried the info app and we get
>>
>> variable_sip_to_user: [011390NNNNNNNNNN;phone-context=+39]
>> variable_sip_to_uri: [011390NNNNNNNNNN;phone-context=+39 at 208.xx.xxx.xxx
>> :5060]
>>
>> Thanks
>> RR
>>
>>
>> On Tue, May 25, 2010 at 4:12 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>>
>>>
>>> On Tue, May 25, 2010 at 12:48 PM, RR <ranjtech at gmail.com> wrote:
>>>
>>>> Hello I want to follow up on this example from YEARS ago. I had tried
>>>> using the variable "destination_number" but that didn't work, and I figured
>>>> that it was because the To: header doesn't have the destination_number but
>>>> has just the URI, so I thought I'd use sip_to_user instead.
>>>>
>>>> We have calls coming in with the following info in the INVITE
>>>>
>>>> From: "16469NNNNNN" <
>>>> sip:16469NNNNNN;phone-context=+1 at 67.1x.xxx.xxx:5060;user=phone
>>>> >;tag=SDru6fc01-gK0c10a887.
>>>> To: <
>>>> sip:011390NNNNNNNNNN;phone-context=+39 at xxx.xxx.xxx.xxx:5060;user=phone
>>>> >.
>>>> (N and x are obviously being masked for privacy)
>>>>
>>>> I use this info in the dialplan like so
>>>>
>>>> <include>
>>>>   <extension name="DIDtest">
>>>>     <condition field="ani" expression="^(\+?|\+1?|1?)(6469NNNNNN).*$"
>>>> break="never">
>>>>         <action application="set" data="effective_caller_id_number=$2"/>
>>>>         <action application="set" data="effective_caller_id_name=$2"/>
>>>>     </condition>
>>>>     <condition field="${sip_to_user}"
>>>> expression="^(\+?|\+1?|1?|011?)(390NNNNNNNNNN).$" break="never">
>>>>         <action application="set" data="continue_on_fail=false"/>
>>>>         <action application="set" data="hangup_after_bridge=true"/>
>>>>         <action application="set" data="domain_name=$${domain}"/>
>>>>         <action application="set" data="bypass_media=true"/>
>>>>         <action application="bridge"
>>>> data="{sip_invite_domain=${sip_from_host}}sofia/gateway/blade2/$2"/>
>>>>     </condition>
>>>>   </extension>
>>>> </include>
>>>>
>>>> However, the calls aren't passing the condition in this dialplan and
>>>> thus not being forwarded to "blade2" server. In fact, even the 011 is not
>>>> being stripped off.
>>>>
>>>> What am I doing wrong?
>>>>
>>>
>>> Create a quick test extension that only does an info dump. (See 9992 in
>>> default.xml for an example.) Make a call, look at the info dump, and make
>>> sure that what you think you are getting is really what you are getting. :)
>>>
>>> -MC
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From phone.bytes at gmail.com  Tue May 25 15:08:36 2010
From: phone.bytes at gmail.com (Phone)
Date: Tue, 25 May 2010 16:08:36 -0600
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
In-Reply-To: <AANLkTikXWvEB_C3jqOWLHaVcns9-U5Hok-5MqgE_uzzZ@mail.gmail.com>
References: <490aedfdd7c8ac680bbe0d7f68c786de@mail.gmail.com>	<4BEB3DD4.9070807@gmail.com>	<EC392AB972A94F8E8D8369C802771451@dell9400>	<4BFBE6C2.5000703@gmail.com>	<AANLkTilsHn9znajzF0Jj-QDu4je9QW7CeymuBYE0SyKa@mail.gmail.com>	<4BFBEF7C.9020107@gmail.com>	<9250A9FF05DB4B879DEF6ECE8E3FFAC1@dell9400>	<AANLkTilU2OwgH0Qk_Qoc3R7ErPO_Z_VRhePusetkRkeW@mail.gmail.com>
	<AANLkTikXWvEB_C3jqOWLHaVcns9-U5Hok-5MqgE_uzzZ@mail.gmail.com>
Message-ID: <4BFC4A64.5090208@gmail.com>

Thanks for the info.  What language did you use?

Bob Coleman wrote:
> Have just recently completed a project to convert an old windows
> dialogic application(vb6) to FreeSWITCH, would reccommend using the
> ESL, was able to map the old dialogic calls to the ESL calls pretty
> easily. We used a mixture of inbound and outbound sockets, as we have
> people dialing us, not just dialing out etc.
>
> With the dialogic you open a port and make the call and handle the
> dtmf, with freeswitch you create a socket connection to FreeSWITCH to
> dial the number and then hand it off to an extension for processing
> the dtmf(that is one approach any way)
>
> Bob
>
> On Wed, May 26, 2010 at 6:45 AM, Michael Collins <msc at freeswitch.org> wrote:
>   
>> On Tue, May 25, 2010 at 11:38 AM, Jan Berger <jan.berger at video24.no> wrote:
>>     
>>> Actually - before you get "to smart" - may I suggest that you start
>>> writing
>>> - or improving - the getting started sections of the doc. Address the
>>> areas
>>> where you struggle and let others benefit from your work.
>>>
>>> I have been through similar issues myself - FS is one of the easier
>>> projects
>>> to work with once you get under the hood, but you basically need to evolve
>>> to the level where you read the source code.
>>>
>>>       
>> And if you can wait 2+ months for "the book" then that should help as well.
>> :D
>> -MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>     
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From anthony.minessale at gmail.com  Tue May 25 15:08:59 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 25 May 2010 17:08:59 -0500
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
In-Reply-To: <AANLkTikXWvEB_C3jqOWLHaVcns9-U5Hok-5MqgE_uzzZ@mail.gmail.com>
References: <490aedfdd7c8ac680bbe0d7f68c786de@mail.gmail.com>
	<4BEB3DD4.9070807@gmail.com>
	<EC392AB972A94F8E8D8369C802771451@dell9400>
	<4BFBE6C2.5000703@gmail.com>
	<AANLkTilsHn9znajzF0Jj-QDu4je9QW7CeymuBYE0SyKa@mail.gmail.com>
	<4BFBEF7C.9020107@gmail.com>
	<9250A9FF05DB4B879DEF6ECE8E3FFAC1@dell9400>
	<AANLkTilU2OwgH0Qk_Qoc3R7ErPO_Z_VRhePusetkRkeW@mail.gmail.com>
	<AANLkTikXWvEB_C3jqOWLHaVcns9-U5Hok-5MqgE_uzzZ@mail.gmail.com>
Message-ID: <AANLkTimju4n8s9rridD_yb7fNXg6dYgX_Gaq7TNQhH56@mail.gmail.com>

If you have anything from that experience you could share in terms of
documentation
or samples, we would welcome it in contrib and/or the wiki.



On Tue, May 25, 2010 at 4:40 PM, Bob Coleman <bobc at devassert.com> wrote:

> Have just recently completed a project to convert an old windows
> dialogic application(vb6) to FreeSWITCH, would reccommend using the
> ESL, was able to map the old dialogic calls to the ESL calls pretty
> easily. We used a mixture of inbound and outbound sockets, as we have
> people dialing us, not just dialing out etc.
>
> With the dialogic you open a port and make the call and handle the
> dtmf, with freeswitch you create a socket connection to FreeSWITCH to
> dial the number and then hand it off to an extension for processing
> the dtmf(that is one approach any way)
>
> Bob
>
> On Wed, May 26, 2010 at 6:45 AM, Michael Collins <msc at freeswitch.org>
> wrote:
> >
> >
> > On Tue, May 25, 2010 at 11:38 AM, Jan Berger <jan.berger at video24.no>
> wrote:
> >>
> >> Actually - before you get "to smart" - may I suggest that you start
> >> writing
> >> - or improving - the getting started sections of the doc. Address the
> >> areas
> >> where you struggle and let others benefit from your work.
> >>
> >> I have been through similar issues myself - FS is one of the easier
> >> projects
> >> to work with once you get under the hood, but you basically need to
> evolve
> >> to the level where you read the source code.
> >>
> > And if you can wait 2+ months for "the book" then that should help as
> well.
> > :D
> > -MC
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <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
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From anthony.minessale at gmail.com  Tue May 25 15:10:54 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 25 May 2010 17:10:54 -0500
Subject: [Freeswitch-users] Direct inward dialling
In-Reply-To: <AANLkTinjUuCyrK6eVSalV6fUVNFM-j_zlvrzRxq3lJZQ@mail.gmail.com>
References: <491B305B.4060307@gmx.net> <491BF6DB.5030406@gmx.net>
	<191c3a030811130621p72a4be5cje130874670a7c714@mail.gmail.com>
	<491C3F16.5010500@gmx.net>
	<AANLkTin9KcsOPtJybqW9tNatRvRUR_2ND7XnecayCstu@mail.gmail.com>
	<AANLkTinPOe6hM0_ZToxxnThCecXCtGyNxtolUkS9w4of@mail.gmail.com>
	<AANLkTimYlmCTFspJo0531NVf5W6QOpJNs_fzFtB8Vi3e@mail.gmail.com>
	<072A175C-9004-4C14-90EC-9A93A8453787@gmail.com>
	<AANLkTinxYIG6S30XMB5GUVuWsXlyhG_slqc_e0PQy0wM@mail.gmail.com>
	<AANLkTinjUuCyrK6eVSalV6fUVNFM-j_zlvrzRxq3lJZQ@mail.gmail.com>
Message-ID: <AANLkTik0JvNDXax6Ay99hrcgzuiItY0f4LkCsaRjYS8a@mail.gmail.com>

you probably have to remove the $ at the end to allow for the ;params

On Tue, May 25, 2010 at 4:43 PM, RR <ranjtech at gmail.com> wrote:

> Hi Anthony,
>
> this is what I see in the debug:
>
> Dialplan: sofia/external/16469NNNNNN Regex (FAIL) [DIDtest]
> ${sip_to_user}(011390NNNNNNNNNN;phone-context=+39) =~
> /^(\+?|\+1?|1?|011?)(390NNNNNNNNNN).$/ break=on-false
>
> and then it moves on to another dialplan xml file.
>
> please note that info app shows:
>
> variable_sip_to_user: [011390NNNNNNNNNN;phone-context=+39]
>
> shouldn't this actually match??
>
> Oh and yes, the copy of FS is pretty old but this is a production system
> which gets 24 / 7 traffic so the upgrade is being pushed and pushed :(
>
> you think this is simply because it's an old build?
>
> Thanks
> RR
>
>
> On Tue, May 25, 2010 at 4:49 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> did you turn up the debug (press f8 or type console loglevel debug)
>> The debug logs will show the data being passed into the regex and the
>> results.
>>
>> P.S.
>> I hope only your example is from years ago and not your copy of FS.
>>
>>
>> On Tue, May 25, 2010 at 3:40 PM, David Ponzone <david.ponzone at gmail.com>wrote:
>>
>>> Which means there is no @ in the sip: part of the SIP To field. Only in
>>> the phone-context part.
>>> FS uses the @ to split the strings into pieces, and then in your case, it
>>> fails as one is missing.
>>>
>>>  David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>> l'exp?diteur.*
>>> *
>>> *
>>>
>>>
>>>
>>> Le 25/05/2010 ? 22:28, RR a ?crit :
>>>
>>> Hi Guys,
>>>
>>> Thanks for the quick feedback
>>>
>>> David, no we're getting the full URI with the domain part intact, just
>>> nothing before the "<" braces
>>>
>>> Michael, I already tried the info app and we get
>>>
>>> variable_sip_to_user: [011390NNNNNNNNNN;phone-context=+39]
>>> variable_sip_to_uri: [011390NNNNNNNNNN;phone-context=+39 at 208.xx.xxx.xxx
>>> :5060]
>>>
>>> Thanks
>>> RR
>>>
>>>
>>> On Tue, May 25, 2010 at 4:12 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>>
>>>>
>>>> On Tue, May 25, 2010 at 12:48 PM, RR <ranjtech at gmail.com> wrote:
>>>>
>>>>> Hello I want to follow up on this example from YEARS ago. I had tried
>>>>> using the variable "destination_number" but that didn't work, and I figured
>>>>> that it was because the To: header doesn't have the destination_number but
>>>>> has just the URI, so I thought I'd use sip_to_user instead.
>>>>>
>>>>> We have calls coming in with the following info in the INVITE
>>>>>
>>>>> From: "16469NNNNNN" <
>>>>> sip:16469NNNNNN;phone-context=+1 at 67.1x.xxx.xxx:5060;user=phone
>>>>> >;tag=SDru6fc01-gK0c10a887.
>>>>> To: <
>>>>> sip:011390NNNNNNNNNN;phone-context=+39 at xxx.xxx.xxx.xxx:5060;user=phone
>>>>> >.
>>>>> (N and x are obviously being masked for privacy)
>>>>>
>>>>> I use this info in the dialplan like so
>>>>>
>>>>> <include>
>>>>>   <extension name="DIDtest">
>>>>>     <condition field="ani" expression="^(\+?|\+1?|1?)(6469NNNNNN).*$"
>>>>> break="never">
>>>>>         <action application="set"
>>>>> data="effective_caller_id_number=$2"/>
>>>>>         <action application="set" data="effective_caller_id_name=$2"/>
>>>>>     </condition>
>>>>>     <condition field="${sip_to_user}"
>>>>> expression="^(\+?|\+1?|1?|011?)(390NNNNNNNNNN).$" break="never">
>>>>>         <action application="set" data="continue_on_fail=false"/>
>>>>>         <action application="set" data="hangup_after_bridge=true"/>
>>>>>         <action application="set" data="domain_name=$${domain}"/>
>>>>>         <action application="set" data="bypass_media=true"/>
>>>>>         <action application="bridge"
>>>>> data="{sip_invite_domain=${sip_from_host}}sofia/gateway/blade2/$2"/>
>>>>>     </condition>
>>>>>   </extension>
>>>>> </include>
>>>>>
>>>>> However, the calls aren't passing the condition in this dialplan and
>>>>> thus not being forwarded to "blade2" server. In fact, even the 011 is not
>>>>> being stripped off.
>>>>>
>>>>> What am I doing wrong?
>>>>>
>>>>
>>>> Create a quick test extension that only does an info dump. (See 9992 in
>>>> default.xml for an example.) Make a call, look at the info dump, and make
>>>> sure that what you think you are getting is really what you are getting. :)
>>>>
>>>> -MC
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>>  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-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
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
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From ranjtech at gmail.com  Tue May 25 15:34:04 2010
From: ranjtech at gmail.com (RR)
Date: Tue, 25 May 2010 18:34:04 -0400
Subject: [Freeswitch-users] Direct inward dialling
In-Reply-To: <AANLkTik0JvNDXax6Ay99hrcgzuiItY0f4LkCsaRjYS8a@mail.gmail.com>
References: <491B305B.4060307@gmx.net>
	<191c3a030811130621p72a4be5cje130874670a7c714@mail.gmail.com>
	<491C3F16.5010500@gmx.net>
	<AANLkTin9KcsOPtJybqW9tNatRvRUR_2ND7XnecayCstu@mail.gmail.com>
	<AANLkTinPOe6hM0_ZToxxnThCecXCtGyNxtolUkS9w4of@mail.gmail.com>
	<AANLkTimYlmCTFspJo0531NVf5W6QOpJNs_fzFtB8Vi3e@mail.gmail.com>
	<072A175C-9004-4C14-90EC-9A93A8453787@gmail.com>
	<AANLkTinxYIG6S30XMB5GUVuWsXlyhG_slqc_e0PQy0wM@mail.gmail.com>
	<AANLkTinjUuCyrK6eVSalV6fUVNFM-j_zlvrzRxq3lJZQ@mail.gmail.com>
	<AANLkTik0JvNDXax6Ay99hrcgzuiItY0f4LkCsaRjYS8a@mail.gmail.com>
Message-ID: <AANLkTinYufK3FWtbM95088fOOOWWzOyMpU6ho0VXiOEy@mail.gmail.com>

Thanks for pointing me in the right direction. What fixed it was adding a
'*' which I used to have before but with all these changes I was making I
forgot to add it back in. So doing this:

<condition field="${sip_to_user}"
expression="^(\+?|\+1?|1?|011?)(390NNNNNNNNNN).*$" break="never">

works
Thanks for the help
\RR


On Tue, May 25, 2010 at 6:10 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> you probably have to remove the $ at the end to allow for the ;params
>
>
> On Tue, May 25, 2010 at 4:43 PM, RR <ranjtech at gmail.com> wrote:
>
>> Hi Anthony,
>>
>> this is what I see in the debug:
>>
>> Dialplan: sofia/external/16469NNNNNN Regex (FAIL) [DIDtest]
>> ${sip_to_user}(011390NNNNNNNNNN;phone-context=+39) =~
>> /^(\+?|\+1?|1?|011?)(390NNNNNNNNNN).$/ break=on-false
>>
>> and then it moves on to another dialplan xml file.
>>
>> please note that info app shows:
>>
>> variable_sip_to_user: [011390NNNNNNNNNN;phone-context=+39]
>>
>> shouldn't this actually match??
>>
>> Oh and yes, the copy of FS is pretty old but this is a production system
>> which gets 24 / 7 traffic so the upgrade is being pushed and pushed :(
>>
>> you think this is simply because it's an old build?
>>
>> Thanks
>> RR
>>
>>
>> On Tue, May 25, 2010 at 4:49 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> did you turn up the debug (press f8 or type console loglevel debug)
>>> The debug logs will show the data being passed into the regex and the
>>> results.
>>>
>>> P.S.
>>> I hope only your example is from years ago and not your copy of FS.
>>>
>>>
>>> On Tue, May 25, 2010 at 3:40 PM, David Ponzone <david.ponzone at gmail.com>wrote:
>>>
>>>> Which means there is no @ in the sip: part of the SIP To field. Only in
>>>> the phone-context part.
>>>> FS uses the @ to split the strings into pieces, and then in your case,
>>>> it fails as one is missing.
>>>>
>>>>  David Ponzone  Direction Technique
>>>> email: david.ponzone at ipeva.fr
>>>> tel:      01 74 03 18 97
>>>> gsm:   06 66 98 76 34
>>>>
>>>> Service Client IPeva
>>>> tel:      0811 46 26 26
>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>>
>>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
>>>> ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>>> l'exp?diteur.*
>>>> *
>>>> *
>>>>
>>>>
>>>>
>>>> Le 25/05/2010 ? 22:28, RR a ?crit :
>>>>
>>>> Hi Guys,
>>>>
>>>> Thanks for the quick feedback
>>>>
>>>> David, no we're getting the full URI with the domain part intact, just
>>>> nothing before the "<" braces
>>>>
>>>> Michael, I already tried the info app and we get
>>>>
>>>> variable_sip_to_user: [011390NNNNNNNNNN;phone-context=+39]
>>>> variable_sip_to_uri: [011390NNNNNNNNNN;phone-context=+39 at 208.xx.xxx.xxx
>>>> :5060]
>>>>
>>>> Thanks
>>>> RR
>>>>
>>>>
>>>> On Tue, May 25, 2010 at 4:12 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>>
>>>>>
>>>>>
>>>>> On Tue, May 25, 2010 at 12:48 PM, RR <ranjtech at gmail.com> wrote:
>>>>>
>>>>>> Hello I want to follow up on this example from YEARS ago. I had tried
>>>>>> using the variable "destination_number" but that didn't work, and I figured
>>>>>> that it was because the To: header doesn't have the destination_number but
>>>>>> has just the URI, so I thought I'd use sip_to_user instead.
>>>>>>
>>>>>> We have calls coming in with the following info in the INVITE
>>>>>>
>>>>>> From: "16469NNNNNN" <
>>>>>> sip:16469NNNNNN;phone-context=+1 at 67.1x.xxx.xxx:5060;user=phone
>>>>>> >;tag=SDru6fc01-gK0c10a887.
>>>>>> To: <
>>>>>> sip:011390NNNNNNNNNN;phone-context=+39 at xxx.xxx.xxx.xxx:5060;user=phone
>>>>>> >.
>>>>>> (N and x are obviously being masked for privacy)
>>>>>>
>>>>>> I use this info in the dialplan like so
>>>>>>
>>>>>> <include>
>>>>>>   <extension name="DIDtest">
>>>>>>     <condition field="ani" expression="^(\+?|\+1?|1?)(6469NNNNNN).*$"
>>>>>> break="never">
>>>>>>         <action application="set"
>>>>>> data="effective_caller_id_number=$2"/>
>>>>>>         <action application="set" data="effective_caller_id_name=$2"/>
>>>>>>     </condition>
>>>>>>     <condition field="${sip_to_user}"
>>>>>> expression="^(\+?|\+1?|1?|011?)(390NNNNNNNNNN).$" break="never">
>>>>>>         <action application="set" data="continue_on_fail=false"/>
>>>>>>         <action application="set" data="hangup_after_bridge=true"/>
>>>>>>         <action application="set" data="domain_name=$${domain}"/>
>>>>>>         <action application="set" data="bypass_media=true"/>
>>>>>>         <action application="bridge"
>>>>>> data="{sip_invite_domain=${sip_from_host}}sofia/gateway/blade2/$2"/>
>>>>>>     </condition>
>>>>>>   </extension>
>>>>>> </include>
>>>>>>
>>>>>> However, the calls aren't passing the condition in this dialplan and
>>>>>> thus not being forwarded to "blade2" server. In fact, even the 011 is not
>>>>>> being stripped off.
>>>>>>
>>>>>> What am I doing wrong?
>>>>>>
>>>>>
>>>>> Create a quick test extension that only does an info dump. (See 9992 in
>>>>> default.xml for an example.) Make a call, look at the info dump, and make
>>>>> sure that what you think you are getting is really what you are getting. :)
>>>>>
>>>>> -MC
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>>  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-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From troy at tlainvestments.com  Tue May 25 17:00:21 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Tue, 25 May 2010 17:00:21 -0700
Subject: [Freeswitch-users] rxgain/txgain dahdi
Message-ID: <B3A254C2-B775-432C-B5D1-768E9BA9E59D@tlainvestments.com>

Ok, I'm googled out!  Can anyone point me to some docs on how to set the rx and tx gain for an FXO or FXS channel using dahdi and mod_openzap?  I think it goes into /etc/dahdi/system.conf, but can't find any docs on how or where.

My /etc/dahdi/system.conf looks like this:

loadzone=us
defaultzone=us
fxsks=1
fxsks=2
fxsks=3
fxsks=4
fxsks=5
fxsks=6
fxoks=7
fxoks=8

Thanks!
-Troy




From msc at freeswitch.org  Tue May 25 17:21:34 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 25 May 2010 17:21:34 -0700
Subject: [Freeswitch-users] rxgain/txgain dahdi
In-Reply-To: <B3A254C2-B775-432C-B5D1-768E9BA9E59D@tlainvestments.com>
References: <B3A254C2-B775-432C-B5D1-768E9BA9E59D@tlainvestments.com>
Message-ID: <AANLkTimxmHgGc1ZpggQcMu11IMVY0ko2Q7tbpAdSG0fB@mail.gmail.com>

Wow, this takes me back. It's been forever. I did find this, but it is
older, zaptelish:
http://www.mattgwatson.ca/2008/05/howto-tune-zaptel-dahdi-fxo-interfaces-on-asterisk-pbx/

Hope it helps...
-MC

On Tue, May 25, 2010 at 5:00 PM, Troy Anderson <troy at tlainvestments.com>wrote:

> Ok, I'm googled out!  Can anyone point me to some docs on how to set the rx
> and tx gain for an FXO or FXS channel using dahdi and mod_openzap?  I think
> it goes into /etc/dahdi/system.conf, but can't find any docs on how or
> where.
>
> My /etc/dahdi/system.conf looks like this:
>
> loadzone=us
> defaultzone=us
> fxsks=1
> fxsks=2
> fxsks=3
> fxsks=4
> fxsks=5
> fxsks=6
> fxoks=7
> fxoks=8
>
> Thanks!
> -Troy
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From ranjtech at gmail.com  Tue May 25 17:27:07 2010
From: ranjtech at gmail.com (RR)
Date: Tue, 25 May 2010 20:27:07 -0400
Subject: [Freeswitch-users] Direct inward dialling
In-Reply-To: <AANLkTinYufK3FWtbM95088fOOOWWzOyMpU6ho0VXiOEy@mail.gmail.com>
References: <491B305B.4060307@gmx.net> <491C3F16.5010500@gmx.net>
	<AANLkTin9KcsOPtJybqW9tNatRvRUR_2ND7XnecayCstu@mail.gmail.com>
	<AANLkTinPOe6hM0_ZToxxnThCecXCtGyNxtolUkS9w4of@mail.gmail.com>
	<AANLkTimYlmCTFspJo0531NVf5W6QOpJNs_fzFtB8Vi3e@mail.gmail.com>
	<072A175C-9004-4C14-90EC-9A93A8453787@gmail.com>
	<AANLkTinxYIG6S30XMB5GUVuWsXlyhG_slqc_e0PQy0wM@mail.gmail.com>
	<AANLkTinjUuCyrK6eVSalV6fUVNFM-j_zlvrzRxq3lJZQ@mail.gmail.com>
	<AANLkTik0JvNDXax6Ay99hrcgzuiItY0f4LkCsaRjYS8a@mail.gmail.com>
	<AANLkTinYufK3FWtbM95088fOOOWWzOyMpU6ho0VXiOEy@mail.gmail.com>
Message-ID: <AANLkTinLTWiKrvgPMiiEZArMxMQFCgtJNXoHO_BBlHRN@mail.gmail.com>

Ok, so I take that back. This seems to only work when the dialplan has a
specific ANI and DNIS / destination_number / sip_to_user defined. If this is
more general

like

<include>
  <extension name="public_did">
    <condition field="ani" expression="^(\+?|\+1?|1?)(\d+).*$"
break="never">
        <action application="set" data="effective_caller_id_number=$2"/>
        <action application="set" data="effective_caller_id_name=$2"/>
    </condition>
    <condition field="${sip_to_user}"
expression="^(\+1?|\+|1?|011?)(\d+).*$" break="never">
      <action application="set" data="continue_on_fail=false"/>
      <action application="set" data="hangup_after_bridge=true"/>
      <action application="set" data="domain_name=$${domain}"/>
      <action application="set" data="bypass_media=true"/>
      <action application="limit_hash" data="in cc_blades 4200 !USER_BUSY"/>
      <action application="bridge"
data="{sip_invite_domain=${sip_from_host}}sofia/gateway/${distributor(cc_blades)}/$2"/>
    </condition>
  </extension>
</include>

then even though the expression/conditions seem to match, none of the digits
are being stripped off. Shouldn't this be stripping off digits??

Here's the debug output:

Dialplan: sofia/external/16469NNNNNN Regex (PASS) [public_did]
ani(16469NNNNNN;phone-context=+1) =~ /^(\+?|\+1?|1?)(\d+).*$/ break=never
Dialplan: sofia/external/16469NNNNNN Action
set(effective_caller_id_number=16469NNNNNN)
Dialplan: sofia/external/16469NNNNNN Action
set(effective_caller_id_name=16469NNNNNN)
Dialplan: sofia/external/16469NNNNNN Regex (PASS) [public_did]
${sip_to_user}(011390NNNNNNNNNN;phone-context=+39) =~
/^(\+1?|\+|1?|011?)(\d+).*$/ break=never
Dialplan: sofia/external/16469NNNNNN Action set(continue_on_fail=false)
Dialplan: sofia/external/16469NNNNNN Action set(hangup_after_bridge=true)
Dialplan: sofia/external/16469NNNNNN Action set(domain_name=208.72.186.166)
Dialplan: sofia/external/16469NNNNNN Action set(bypass_media=true)
Dialplan: sofia/external/16469NNNNNN Action limit_hash(in cc_blades 4200
!USER_BUSY)
Dialplan: sofia/external/16469NNNNNN Action
bridge({sip_invite_domain=${sip_from_host}}sofia/gateway/${distributor(cc_blades)}/011390NNNNNNNNNN)

why're the '1' in the ANI and '011' in the DNIS/sip_to_user being stripped
off???

Thanks in Advance,
RR


On Tue, May 25, 2010 at 6:34 PM, RR <ranjtech at gmail.com> wrote:

> Thanks for pointing me in the right direction. What fixed it was adding a
> '*' which I used to have before but with all these changes I was making I
> forgot to add it back in. So doing this:
>
> <condition field="${sip_to_user}"
> expression="^(\+?|\+1?|1?|011?)(390NNNNNNNNNN).*$" break="never">
>
> works
> Thanks for the help
> \RR
>
>
> On Tue, May 25, 2010 at 6:10 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> you probably have to remove the $ at the end to allow for the ;params
>>
>>
>> On Tue, May 25, 2010 at 4:43 PM, RR <ranjtech at gmail.com> wrote:
>>
>>> Hi Anthony,
>>>
>>> this is what I see in the debug:
>>>
>>> Dialplan: sofia/external/16469NNNNNN Regex (FAIL) [DIDtest]
>>> ${sip_to_user}(011390NNNNNNNNNN;phone-context=+39) =~
>>> /^(\+?|\+1?|1?|011?)(390NNNNNNNNNN).$/ break=on-false
>>>
>>> and then it moves on to another dialplan xml file.
>>>
>>> please note that info app shows:
>>>
>>> variable_sip_to_user: [011390NNNNNNNNNN;phone-context=+39]
>>>
>>> shouldn't this actually match??
>>>
>>> Oh and yes, the copy of FS is pretty old but this is a production system
>>> which gets 24 / 7 traffic so the upgrade is being pushed and pushed :(
>>>
>>> you think this is simply because it's an old build?
>>>
>>> Thanks
>>> RR
>>>
>>>
>>> On Tue, May 25, 2010 at 4:49 PM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> did you turn up the debug (press f8 or type console loglevel debug)
>>>> The debug logs will show the data being passed into the regex and the
>>>> results.
>>>>
>>>> P.S.
>>>> I hope only your example is from years ago and not your copy of FS.
>>>>
>>>>
>>>> On Tue, May 25, 2010 at 3:40 PM, David Ponzone <david.ponzone at gmail.com
>>>> > wrote:
>>>>
>>>>> Which means there is no @ in the sip: part of the SIP To field. Only in
>>>>> the phone-context part.
>>>>> FS uses the @ to split the strings into pieces, and then in your case,
>>>>> it fails as one is missing.
>>>>>
>>>>>  David Ponzone  Direction Technique
>>>>> email: david.ponzone at ipeva.fr
>>>>> tel:      01 74 03 18 97
>>>>> gsm:   06 66 98 76 34
>>>>>
>>>>> Service Client IPeva
>>>>> tel:      0811 46 26 26
>>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>>>
>>>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
>>>>> ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>>>> l'exp?diteur.*
>>>>> *
>>>>> *
>>>>>
>>>>>
>>>>>
>>>>> Le 25/05/2010 ? 22:28, RR a ?crit :
>>>>>
>>>>> Hi Guys,
>>>>>
>>>>> Thanks for the quick feedback
>>>>>
>>>>> David, no we're getting the full URI with the domain part intact, just
>>>>> nothing before the "<" braces
>>>>>
>>>>> Michael, I already tried the info app and we get
>>>>>
>>>>> variable_sip_to_user: [011390NNNNNNNNNN;phone-context=+39]
>>>>> variable_sip_to_uri: [011390NNNNNNNNNN;
>>>>> phone-context=+39 at 208.xx.xxx.xxx:5060]
>>>>>
>>>>> Thanks
>>>>> RR
>>>>>
>>>>>
>>>>> On Tue, May 25, 2010 at 4:12 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>>>
>>>>>>
>>>>>>
>>>>>> On Tue, May 25, 2010 at 12:48 PM, RR <ranjtech at gmail.com> wrote:
>>>>>>
>>>>>>> Hello I want to follow up on this example from YEARS ago. I had tried
>>>>>>> using the variable "destination_number" but that didn't work, and I figured
>>>>>>> that it was because the To: header doesn't have the destination_number but
>>>>>>> has just the URI, so I thought I'd use sip_to_user instead.
>>>>>>>
>>>>>>> We have calls coming in with the following info in the INVITE
>>>>>>>
>>>>>>> From: "16469NNNNNN" <
>>>>>>> sip:16469NNNNNN;phone-context=+1 at 67.1x.xxx.xxx:5060;user=phone
>>>>>>> >;tag=SDru6fc01-gK0c10a887.
>>>>>>> To: <
>>>>>>> sip:011390NNNNNNNNNN;phone-context=+39 at xxx.xxx.xxx.xxx:5060;user=phone
>>>>>>> >.
>>>>>>> (N and x are obviously being masked for privacy)
>>>>>>>
>>>>>>> I use this info in the dialplan like so
>>>>>>>
>>>>>>> <include>
>>>>>>>   <extension name="DIDtest">
>>>>>>>     <condition field="ani" expression="^(\+?|\+1?|1?)(6469NNNNNN).*$"
>>>>>>> break="never">
>>>>>>>         <action application="set"
>>>>>>> data="effective_caller_id_number=$2"/>
>>>>>>>         <action application="set"
>>>>>>> data="effective_caller_id_name=$2"/>
>>>>>>>     </condition>
>>>>>>>     <condition field="${sip_to_user}"
>>>>>>> expression="^(\+?|\+1?|1?|011?)(390NNNNNNNNNN).$" break="never">
>>>>>>>         <action application="set" data="continue_on_fail=false"/>
>>>>>>>         <action application="set" data="hangup_after_bridge=true"/>
>>>>>>>         <action application="set" data="domain_name=$${domain}"/>
>>>>>>>         <action application="set" data="bypass_media=true"/>
>>>>>>>         <action application="bridge"
>>>>>>> data="{sip_invite_domain=${sip_from_host}}sofia/gateway/blade2/$2"/>
>>>>>>>     </condition>
>>>>>>>   </extension>
>>>>>>> </include>
>>>>>>>
>>>>>>> However, the calls aren't passing the condition in this dialplan and
>>>>>>> thus not being forwarded to "blade2" server. In fact, even the 011 is not
>>>>>>> being stripped off.
>>>>>>>
>>>>>>> What am I doing wrong?
>>>>>>>
>>>>>>
>>>>>> Create a quick test extension that only does an info dump. (See 9992
>>>>>> in default.xml for an example.) Make a call, look at the info dump, and make
>>>>>> sure that what you think you are getting is really what you are getting. :)
>>>>>>
>>>>>> -MC
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>>  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-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> 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-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From msc at freeswitch.org  Tue May 25 17:34:46 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 25 May 2010 17:34:46 -0700
Subject: [Freeswitch-users] Direct inward dialling
In-Reply-To: <AANLkTinLTWiKrvgPMiiEZArMxMQFCgtJNXoHO_BBlHRN@mail.gmail.com>
References: <491B305B.4060307@gmx.net>
	<AANLkTin9KcsOPtJybqW9tNatRvRUR_2ND7XnecayCstu@mail.gmail.com>
	<AANLkTinPOe6hM0_ZToxxnThCecXCtGyNxtolUkS9w4of@mail.gmail.com>
	<AANLkTimYlmCTFspJo0531NVf5W6QOpJNs_fzFtB8Vi3e@mail.gmail.com>
	<072A175C-9004-4C14-90EC-9A93A8453787@gmail.com>
	<AANLkTinxYIG6S30XMB5GUVuWsXlyhG_slqc_e0PQy0wM@mail.gmail.com>
	<AANLkTinjUuCyrK6eVSalV6fUVNFM-j_zlvrzRxq3lJZQ@mail.gmail.com>
	<AANLkTik0JvNDXax6Ay99hrcgzuiItY0f4LkCsaRjYS8a@mail.gmail.com>
	<AANLkTinYufK3FWtbM95088fOOOWWzOyMpU6ho0VXiOEy@mail.gmail.com>
	<AANLkTinLTWiKrvgPMiiEZArMxMQFCgtJNXoHO_BBlHRN@mail.gmail.com>
Message-ID: <AANLkTilq_gLLPa-VaRlaiKofqNVlpvm1zgSiL53UTf7z@mail.gmail.com>

On Tue, May 25, 2010 at 5:27 PM, RR <ranjtech at gmail.com> wrote:

> Ok, so I take that back. This seems to only work when the dialplan has a
> specific ANI and DNIS / destination_number / sip_to_user defined. If this is
> more general
>
> like
>
> <include>
>   <extension name="public_did">
>     <condition field="ani" expression="^(\+?|\+1?|1?)(\d+).*$"
> break="never">
>         <action application="set" data="effective_caller_id_number=$2"/>
>         <action application="set" data="effective_caller_id_name=$2"/>
>     </condition>
>     <condition field="${sip_to_user}"
> expression="^(\+1?|\+|1?|011?)(\d+).*$" break="never">
>       <action application="set" data="continue_on_fail=false"/>
>       <action application="set" data="hangup_after_bridge=true"/>
>       <action application="set" data="domain_name=$${domain}"/>
>       <action application="set" data="bypass_media=true"/>
>       <action application="limit_hash" data="in cc_blades 4200
> !USER_BUSY"/>
>       <action application="bridge"
> data="{sip_invite_domain=${sip_from_host}}sofia/gateway/${distributor(cc_blades)}/$2"/>
>     </condition>
>   </extension>
> </include>
>
> then even though the expression/conditions seem to match, none of the
> digits are being stripped off. Shouldn't this be stripping off digits??
>
> Here's the debug output:
>
> Dialplan: sofia/external/16469NNNNNN Regex (PASS) [public_did]
> ani(16469NNNNNN;phone-context=+1) =~ /^(\+?|\+1?|1?)(\d+).*$/ break=never
> Dialplan: sofia/external/16469NNNNNN Action
> set(effective_caller_id_number=16469NNNNNN)
> Dialplan: sofia/external/16469NNNNNN Action
> set(effective_caller_id_name=16469NNNNNN)
> Dialplan: sofia/external/16469NNNNNN Regex (PASS) [public_did]
> ${sip_to_user}(011390NNNNNNNNNN;phone-context=+39) =~
> /^(\+1?|\+|1?|011?)(\d+).*$/ break=never
> Dialplan: sofia/external/16469NNNNNN Action set(continue_on_fail=false)
> Dialplan: sofia/external/16469NNNNNN Action set(hangup_after_bridge=true)
> Dialplan: sofia/external/16469NNNNNN Action set(domain_name=208.72.186.166)
> Dialplan: sofia/external/16469NNNNNN Action set(bypass_media=true)
> Dialplan: sofia/external/16469NNNNNN Action limit_hash(in cc_blades 4200
> !USER_BUSY)
> Dialplan: sofia/external/16469NNNNNN Action
> bridge({sip_invite_domain=${sip_from_host}}sofia/gateway/${distributor(cc_blades)}/011390NNNNNNNNNN)
>
> why're the '1' in the ANI and '011' in the DNIS/sip_to_user being stripped
> off???
>

Regex 101 :)

The 1 or the 011 are in $1
-MC
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From bwibowo at gmail.com  Tue May 25 17:43:10 2010
From: bwibowo at gmail.com (Budi wibowo)
Date: Wed, 26 May 2010 00:43:10 +0000
Subject: [Freeswitch-users] G723 codec
Message-ID: <1055665185-1274834591-cardhu_decombobulator_blackberry.rim.net-1455303023-@bda057.bisx.prodap.on.blackberry>

Hi, is it g723 in fs? Or just pass through?
Other idea regarding codec issue is put transcoding box outside fs box.  You will also have flexibility for conncted client.
Any client any codec can connect 

Br
Budi
------Original Message------
From: Steve Underwood
Sender: freeswitch-users-bounces at lists.freeswitch.org
To: freeswitch-users at lists.freeswitch.org
ReplyTo: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] G723 codec
Sent: May 26, 2010 01:07

On 05/26/2010 01:42 AM, Madovsky wrote:
> Hi all,
> is there a chance that in future there will be a mod_g723 transcoder ?
>
I assume you mean G.723.1. It is very difficult to licence the patents 
related to G.723.1 unless you are a huge user. G.729 has a well defined 
scheme for licencing the relevant patents on an incremental basis, 
starting at a modest numbers of channels. G.723.1 is quite problematic. 
For that reason it is unlikely that G.723.1 will be available as a host 
software transcoder module any time soon. There are, however, hardware 
card options which can provide this codec.

Steve


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


From ranjtech at gmail.com  Tue May 25 17:44:02 2010
From: ranjtech at gmail.com (RR)
Date: Tue, 25 May 2010 20:44:02 -0400
Subject: [Freeswitch-users] Direct inward dialling
In-Reply-To: <AANLkTilq_gLLPa-VaRlaiKofqNVlpvm1zgSiL53UTf7z@mail.gmail.com>
References: <491B305B.4060307@gmx.net>
	<AANLkTinPOe6hM0_ZToxxnThCecXCtGyNxtolUkS9w4of@mail.gmail.com>
	<AANLkTimYlmCTFspJo0531NVf5W6QOpJNs_fzFtB8Vi3e@mail.gmail.com>
	<072A175C-9004-4C14-90EC-9A93A8453787@gmail.com>
	<AANLkTinxYIG6S30XMB5GUVuWsXlyhG_slqc_e0PQy0wM@mail.gmail.com>
	<AANLkTinjUuCyrK6eVSalV6fUVNFM-j_zlvrzRxq3lJZQ@mail.gmail.com>
	<AANLkTik0JvNDXax6Ay99hrcgzuiItY0f4LkCsaRjYS8a@mail.gmail.com>
	<AANLkTinYufK3FWtbM95088fOOOWWzOyMpU6ho0VXiOEy@mail.gmail.com>
	<AANLkTinLTWiKrvgPMiiEZArMxMQFCgtJNXoHO_BBlHRN@mail.gmail.com>
	<AANLkTilq_gLLPa-VaRlaiKofqNVlpvm1zgSiL53UTf7z@mail.gmail.com>
Message-ID: <AANLkTikiraunYdhjcCUmKLmgNAyQ-Sp6r4EpohHuEbZn@mail.gmail.com>

Michael, haha, yeah they indeed are. That's why I'm routing based on $2, but
I still see the 1 and/or the 011 going through to the "bridge" application.
Why??

On Tue, May 25, 2010 at 8:34 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Tue, May 25, 2010 at 5:27 PM, RR <ranjtech at gmail.com> wrote:
>
>> Ok, so I take that back. This seems to only work when the dialplan has a
>> specific ANI and DNIS / destination_number / sip_to_user defined. If this is
>> more general
>>
>> like
>>
>> <include>
>>   <extension name="public_did">
>>     <condition field="ani" expression="^(\+?|\+1?|1?)(\d+).*$"
>> break="never">
>>         <action application="set" data="effective_caller_id_number=$2"/>
>>         <action application="set" data="effective_caller_id_name=$2"/>
>>     </condition>
>>     <condition field="${sip_to_user}"
>> expression="^(\+1?|\+|1?|011?)(\d+).*$" break="never">
>>       <action application="set" data="continue_on_fail=false"/>
>>       <action application="set" data="hangup_after_bridge=true"/>
>>       <action application="set" data="domain_name=$${domain}"/>
>>       <action application="set" data="bypass_media=true"/>
>>       <action application="limit_hash" data="in cc_blades 4200
>> !USER_BUSY"/>
>>       <action application="bridge"
>> data="{sip_invite_domain=${sip_from_host}}sofia/gateway/${distributor(cc_blades)}/$2"/>
>>     </condition>
>>   </extension>
>> </include>
>>
>> then even though the expression/conditions seem to match, none of the
>> digits are being stripped off. Shouldn't this be stripping off digits??
>>
>> Here's the debug output:
>>
>> Dialplan: sofia/external/16469NNNNNN Regex (PASS) [public_did]
>> ani(16469NNNNNN;phone-context=+1) =~ /^(\+?|\+1?|1?)(\d+).*$/ break=never
>> Dialplan: sofia/external/16469NNNNNN Action
>> set(effective_caller_id_number=16469NNNNNN)
>> Dialplan: sofia/external/16469NNNNNN Action
>> set(effective_caller_id_name=16469NNNNNN)
>> Dialplan: sofia/external/16469NNNNNN Regex (PASS) [public_did]
>> ${sip_to_user}(011390NNNNNNNNNN;phone-context=+39) =~
>> /^(\+1?|\+|1?|011?)(\d+).*$/ break=never
>> Dialplan: sofia/external/16469NNNNNN Action set(continue_on_fail=false)
>> Dialplan: sofia/external/16469NNNNNN Action set(hangup_after_bridge=true)
>> Dialplan: sofia/external/16469NNNNNN Action
>> set(domain_name=208.72.186.166)
>> Dialplan: sofia/external/16469NNNNNN Action set(bypass_media=true)
>> Dialplan: sofia/external/16469NNNNNN Action limit_hash(in cc_blades 4200
>> !USER_BUSY)
>> Dialplan: sofia/external/16469NNNNNN Action
>> bridge({sip_invite_domain=${sip_from_host}}sofia/gateway/${distributor(cc_blades)}/011390NNNNNNNNNN)
>>
>> why're the '1' in the ANI and '011' in the DNIS/sip_to_user being stripped
>> off???
>>
>
> Regex 101 :)
>
> The 1 or the 011 are in $1
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dujinfang at gmail.com  Tue May 25 17:48:18 2010
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 26 May 2010 08:48:18 +0800
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
In-Reply-To: <AANLkTimju4n8s9rridD_yb7fNXg6dYgX_Gaq7TNQhH56@mail.gmail.com>
References: <490aedfdd7c8ac680bbe0d7f68c786de@mail.gmail.com>
	<4BEB3DD4.9070807@gmail.com>
	<EC392AB972A94F8E8D8369C802771451@dell9400>
	<4BFBE6C2.5000703@gmail.com>
	<AANLkTilsHn9znajzF0Jj-QDu4je9QW7CeymuBYE0SyKa@mail.gmail.com>
	<4BFBEF7C.9020107@gmail.com>
	<9250A9FF05DB4B879DEF6ECE8E3FFAC1@dell9400>
	<AANLkTilU2OwgH0Qk_Qoc3R7ErPO_Z_VRhePusetkRkeW@mail.gmail.com>
	<AANLkTikXWvEB_C3jqOWLHaVcns9-U5Hok-5MqgE_uzzZ@mail.gmail.com>
	<AANLkTimju4n8s9rridD_yb7fNXg6dYgX_Gaq7TNQhH56@mail.gmail.com>
Message-ID: <AANLkTim52aA7EW2UkIqLmdx-qZN2ZbNoCcl0AaI4Su3T@mail.gmail.com>

Another option is Erlang. I'd like to share my experience and also
comments are welcome.

http://www.dujinfang.com/past/2010/4/22/build-a-complex-hence-powerful-freeswitch-ivr-in-erlang/

2010/5/26 Anthony Minessale <anthony.minessale at gmail.com>:
> If you have anything from that experience you could share in terms of
> documentation
> or samples, we would welcome it in contrib and/or the wiki.
>
>
> On Tue, May 25, 2010 at 4:40 PM, Bob Coleman <bobc at devassert.com> wrote:
>>
>> Have just recently completed a project to convert an old windows
>> dialogic application(vb6) to FreeSWITCH, would reccommend using the
>> ESL, was able to map the old dialogic calls to the ESL calls pretty
>> easily. We used a mixture of inbound and outbound sockets, as we have
>> people dialing us, not just dialing out etc.
>>
>> With the dialogic you open a port and make the call and handle the
>> dtmf, with freeswitch you create a socket connection to FreeSWITCH to
>> dial the number and then hand it off to an extension for processing
>> the dtmf(that is one approach any way)
>>
>> Bob
>>
>> On Wed, May 26, 2010 at 6:45 AM, Michael Collins <msc at freeswitch.org>
>> wrote:
>> >
>> >
>> > On Tue, May 25, 2010 at 11:38 AM, Jan Berger <jan.berger at video24.no>
>> > wrote:
>> >>
>> >> Actually - before you get "to smart" - may I suggest that you start
>> >> writing
>> >> - or improving - the getting started sections of the doc. Address the
>> >> areas
>> >> where you struggle and let others benefit from your work.
>> >>
>> >> I have been through similar issues myself - FS is one of the easier
>> >> projects
>> >> to work with once you get under the hood, but you basically need to
>> >> evolve
>> >> to the level where you read the source code.
>> >>
>> > And if you can wait 2+ months for "the book" then that should help as
>> > well.
>> > :D
>> > -MC
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



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


From msc at freeswitch.org  Tue May 25 18:19:36 2010
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 25 May 2010 18:19:36 -0700
Subject: [Freeswitch-users] Direct inward dialling
In-Reply-To: <AANLkTikiraunYdhjcCUmKLmgNAyQ-Sp6r4EpohHuEbZn@mail.gmail.com>
References: <491B305B.4060307@gmx.net>
	<AANLkTimYlmCTFspJo0531NVf5W6QOpJNs_fzFtB8Vi3e@mail.gmail.com>
	<072A175C-9004-4C14-90EC-9A93A8453787@gmail.com>
	<AANLkTinxYIG6S30XMB5GUVuWsXlyhG_slqc_e0PQy0wM@mail.gmail.com>
	<AANLkTinjUuCyrK6eVSalV6fUVNFM-j_zlvrzRxq3lJZQ@mail.gmail.com>
	<AANLkTik0JvNDXax6Ay99hrcgzuiItY0f4LkCsaRjYS8a@mail.gmail.com>
	<AANLkTinYufK3FWtbM95088fOOOWWzOyMpU6ho0VXiOEy@mail.gmail.com>
	<AANLkTinLTWiKrvgPMiiEZArMxMQFCgtJNXoHO_BBlHRN@mail.gmail.com>
	<AANLkTilq_gLLPa-VaRlaiKofqNVlpvm1zgSiL53UTf7z@mail.gmail.com>
	<AANLkTikiraunYdhjcCUmKLmgNAyQ-Sp6r4EpohHuEbZn@mail.gmail.com>
Message-ID: <AANLkTimMkeVcO7mqzCVA95PE_XSGoGdHfC2A7HJ5i0_R@mail.gmail.com>

On Tue, May 25, 2010 at 5:44 PM, RR <ranjtech at gmail.com> wrote:

> Michael, haha, yeah they indeed are. That's why I'm routing based on $2,
> but I still see the 1 and/or the 011 going through to the "bridge"
> application. Why??
>
Because your regex is wrong. :) It took me a while to figure it out. I'm
surprised it worked at all. All the stuff you have inside the first set of
parens is not behaving the way you think it should be. If I read your
intentions correctly you're trying to strip off leading:
+
OR
+1
OR
1

In the first regex. Correct? If ANI is NANPA-ish then try this in your first
regex:
^\+?1?([2-9]\d+).*$

That should strip off leading + and/or 1 and capture just the 10-digit phone
number in $1. (Be sure to use $1 and not $2, unless you had your heart set
on using $2 in which case wrap the first part of the regex in parens)

The other regex is also tricky. I assume you are trying to strip off the
same as above as well as 011? Try this:
^\+?1?(011)?([2-9]\d+).*$

Again, if the phone number in question is NANPA then $2 should contain just
the 10 digits you want. Play around with that and let us know what happens.
Also, don't forget what I said about using regex from the fs_cli. You can
test all this stuff yourself. :)

-MC




>
> On Tue, May 25, 2010 at 8:34 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Tue, May 25, 2010 at 5:27 PM, RR <ranjtech at gmail.com> wrote:
>>
>>> Ok, so I take that back. This seems to only work when the dialplan has a
>>> specific ANI and DNIS / destination_number / sip_to_user defined. If this is
>>> more general
>>>
>>> like
>>>
>>> <include>
>>>   <extension name="public_did">
>>>     <condition field="ani" expression="^(\+?|\+1?|1?)(\d+).*$"
>>> break="never">
>>>         <action application="set" data="effective_caller_id_number=$2"/>
>>>         <action application="set" data="effective_caller_id_name=$2"/>
>>>     </condition>
>>>     <condition field="${sip_to_user}"
>>> expression="^(\+1?|\+|1?|011?)(\d+).*$" break="never">
>>>       <action application="set" data="continue_on_fail=false"/>
>>>       <action application="set" data="hangup_after_bridge=true"/>
>>>       <action application="set" data="domain_name=$${domain}"/>
>>>       <action application="set" data="bypass_media=true"/>
>>>       <action application="limit_hash" data="in cc_blades 4200
>>> !USER_BUSY"/>
>>>       <action application="bridge"
>>> data="{sip_invite_domain=${sip_from_host}}sofia/gateway/${distributor(cc_blades)}/$2"/>
>>>     </condition>
>>>   </extension>
>>> </include>
>>>
>>> then even though the expression/conditions seem to match, none of the
>>> digits are being stripped off. Shouldn't this be stripping off digits??
>>>
>>> Here's the debug output:
>>>
>>> Dialplan: sofia/external/16469NNNNNN Regex (PASS) [public_did]
>>> ani(16469NNNNNN;phone-context=+1) =~ /^(\+?|\+1?|1?)(\d+).*$/ break=never
>>> Dialplan: sofia/external/16469NNNNNN Action
>>> set(effective_caller_id_number=16469NNNNNN)
>>> Dialplan: sofia/external/16469NNNNNN Action
>>> set(effective_caller_id_name=16469NNNNNN)
>>> Dialplan: sofia/external/16469NNNNNN Regex (PASS) [public_did]
>>> ${sip_to_user}(011390NNNNNNNNNN;phone-context=+39) =~
>>> /^(\+1?|\+|1?|011?)(\d+).*$/ break=never
>>> Dialplan: sofia/external/16469NNNNNN Action set(continue_on_fail=false)
>>> Dialplan: sofia/external/16469NNNNNN Action set(hangup_after_bridge=true)
>>> Dialplan: sofia/external/16469NNNNNN Action
>>> set(domain_name=208.72.186.166)
>>> Dialplan: sofia/external/16469NNNNNN Action set(bypass_media=true)
>>> Dialplan: sofia/external/16469NNNNNN Action limit_hash(in cc_blades 4200
>>> !USER_BUSY)
>>> Dialplan: sofia/external/16469NNNNNN Action
>>> bridge({sip_invite_domain=${sip_from_host}}sofia/gateway/${distributor(cc_blades)}/011390NNNNNNNNNN)
>>>
>>> why're the '1' in the ANI and '011' in the DNIS/sip_to_user being
>>> stripped off???
>>>
>>
>> Regex 101 :)
>>
>> The 1 or the 011 are in $1
>> -MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bobc at devassert.com  Tue May 25 19:07:12 2010
From: bobc at devassert.com (Bob Coleman)
Date: Wed, 26 May 2010 14:07:12 +1200
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
In-Reply-To: <4BFC4A64.5090208@gmail.com>
References: <490aedfdd7c8ac680bbe0d7f68c786de@mail.gmail.com>
	<4BEB3DD4.9070807@gmail.com>
	<EC392AB972A94F8E8D8369C802771451@dell9400>
	<4BFBE6C2.5000703@gmail.com>
	<AANLkTilsHn9znajzF0Jj-QDu4je9QW7CeymuBYE0SyKa@mail.gmail.com>
	<4BFBEF7C.9020107@gmail.com>
	<9250A9FF05DB4B879DEF6ECE8E3FFAC1@dell9400>
	<AANLkTilU2OwgH0Qk_Qoc3R7ErPO_Z_VRhePusetkRkeW@mail.gmail.com>
	<AANLkTikXWvEB_C3jqOWLHaVcns9-U5Hok-5MqgE_uzzZ@mail.gmail.com>
	<4BFC4A64.5090208@gmail.com>
Message-ID: <AANLkTinFtVXa7CPou-ShPYIXHq0AktokhptGSEFKBkge@mail.gmail.com>

We used c# as the rest of our systems are windows based. The language
doesnt matter too much, as long as you know where you are headed, what
performance you require, and what platform you are going to be using.

Found the ESL so much easier than the dialogic c library we were using.

The docs for the esl are easy to understand, the thing I couldnt get
my head around initially was the dialing out, with the dialogic you
are in the middle when you dial, ie already on the channel, but with
freeswitch you are kind of the third party when you dial, the channel
being created by the dialing and handing it off to be worked on. We
make the call via an inbound event socket and hand it off to an
outbound event socket application via the dialplan.

On Wed, May 26, 2010 at 10:08 AM, Phone <phone.bytes at gmail.com> wrote:
> Thanks for the info. ?What language did you use?
>
> Bob Coleman wrote:
>> Have just recently completed a project to convert an old windows
>> dialogic application(vb6) to FreeSWITCH, would reccommend using the
>> ESL, was able to map the old dialogic calls to the ESL calls pretty
>> easily. We used a mixture of inbound and outbound sockets, as we have
>> people dialing us, not just dialing out etc.
>>
>> With the dialogic you open a port and make the call and handle the
>> dtmf, with freeswitch you create a socket connection to FreeSWITCH to
>> dial the number and then hand it off to an extension for processing
>> the dtmf(that is one approach any way)
>>
>> Bob
>>
>> On Wed, May 26, 2010 at 6:45 AM, Michael Collins <msc at freeswitch.org> wrote:
>>
>>> On Tue, May 25, 2010 at 11:38 AM, Jan Berger <jan.berger at video24.no> wrote:
>>>
>>>> Actually - before you get "to smart" - may I suggest that you start
>>>> writing
>>>> - or improving - the getting started sections of the doc. Address the
>>>> areas
>>>> where you struggle and let others benefit from your work.
>>>>
>>>> I have been through similar issues myself - FS is one of the easier
>>>> projects
>>>> to work with once you get under the hood, but you basically need to evolve
>>>> to the level where you read the source code.
>>>>
>>>>
>>> And if you can wait 2+ months for "the book" then that should help as well.
>>> :D
>>> -MC
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From bobc at devassert.com  Tue May 25 19:23:44 2010
From: bobc at devassert.com (Bob Coleman)
Date: Wed, 26 May 2010 14:23:44 +1200
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
In-Reply-To: <AANLkTimju4n8s9rridD_yb7fNXg6dYgX_Gaq7TNQhH56@mail.gmail.com>
References: <490aedfdd7c8ac680bbe0d7f68c786de@mail.gmail.com>
	<4BEB3DD4.9070807@gmail.com>
	<EC392AB972A94F8E8D8369C802771451@dell9400>
	<4BFBE6C2.5000703@gmail.com>
	<AANLkTilsHn9znajzF0Jj-QDu4je9QW7CeymuBYE0SyKa@mail.gmail.com>
	<4BFBEF7C.9020107@gmail.com>
	<9250A9FF05DB4B879DEF6ECE8E3FFAC1@dell9400>
	<AANLkTilU2OwgH0Qk_Qoc3R7ErPO_Z_VRhePusetkRkeW@mail.gmail.com>
	<AANLkTikXWvEB_C3jqOWLHaVcns9-U5Hok-5MqgE_uzzZ@mail.gmail.com>
	<AANLkTimju4n8s9rridD_yb7fNXg6dYgX_Gaq7TNQhH56@mail.gmail.com>
Message-ID: <AANLkTilgMFjtnSGGVKdVulz1FmTuJDk9rcisK7IheguT@mail.gmail.com>

One thing I did find useful coming from the dialogic conversion was
the playback_terminators variable, which Anthony helped me find. This
allowed us to trap what the user to started to enter while a prompt
was being played and then continue to accept input via other commands,
and gives the application an async feel, similar to the dialogic, with
out having to code against async outbound event socket.

On Wed, May 26, 2010 at 10:08 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> If you have anything from that experience you could share in terms of
> documentation
> or samples, we would welcome it in contrib and/or the wiki.
>
>
> On Tue, May 25, 2010 at 4:40 PM, Bob Coleman <bobc at devassert.com> wrote:
>>
>> Have just recently completed a project to convert an old windows
>> dialogic application(vb6) to FreeSWITCH, would reccommend using the
>> ESL, was able to map the old dialogic calls to the ESL calls pretty
>> easily. We used a mixture of inbound and outbound sockets, as we have
>> people dialing us, not just dialing out etc.
>>
>> With the dialogic you open a port and make the call and handle the
>> dtmf, with freeswitch you create a socket connection to FreeSWITCH to
>> dial the number and then hand it off to an extension for processing
>> the dtmf(that is one approach any way)
>>
>> Bob
>>
>> On Wed, May 26, 2010 at 6:45 AM, Michael Collins <msc at freeswitch.org>
>> wrote:
>> >
>> >
>> > On Tue, May 25, 2010 at 11:38 AM, Jan Berger <jan.berger at video24.no>
>> > wrote:
>> >>
>> >> Actually - before you get "to smart" - may I suggest that you start
>> >> writing
>> >> - or improving - the getting started sections of the doc. Address the
>> >> areas
>> >> where you struggle and let others benefit from your work.
>> >>
>> >> I have been through similar issues myself - FS is one of the easier
>> >> projects
>> >> to work with once you get under the hood, but you basically need to
>> >> evolve
>> >> to the level where you read the source code.
>> >>
>> > And if you can wait 2+ months for "the book" then that should help as
>> > well.
>> > :D
>> > -MC
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From jan.berger at video24.no  Tue May 25 19:26:43 2010
From: jan.berger at video24.no (Jan Berger)
Date: Wed, 26 May 2010 04:26:43 +0200
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
In-Reply-To: <AANLkTinFtVXa7CPou-ShPYIXHq0AktokhptGSEFKBkge@mail.gmail.com>
References: <490aedfdd7c8ac680bbe0d7f68c786de@mail.gmail.com>
	<4BEB3DD4.9070807@gmail.com>
	<EC392AB972A94F8E8D8369C802771451@dell9400>
	<4BFBE6C2.5000703@gmail.com>
	<AANLkTilsHn9znajzF0Jj-QDu4je9QW7CeymuBYE0SyKa@mail.gmail.com>
	<4BFBEF7C.9020107@gmail.com>
	<9250A9FF05DB4B879DEF6ECE8E3FFAC1@dell9400>
	<AANLkTilU2OwgH0Qk_Qoc3R7ErPO_Z_VRhePusetkRkeW@mail.gmail.com>
	<AANLkTikXWvEB_C3jqOWLHaVcns9-U5Hok-5MqgE_uzzZ@mail.gmail.com>
	<4BFC4A64.5090208@gmail.com>
	<AANLkTinFtVXa7CPou-ShPYIXHq0AktokhptGSEFKBkge@mail.gmail.com>
Message-ID: <D311BDCB06324D0DB77898FC6FF4B0EB@dell9400>

Hi,

Do you have some sample code you could share + what docs did you look at?

I would like to write and test some C# using ESL for my own work.

Jan

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bob
Coleman
Sent: 26. mai 2010 04:07
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Questions on Building an application for
FreeSWITCH

We used c# as the rest of our systems are windows based. The language
doesnt matter too much, as long as you know where you are headed, what
performance you require, and what platform you are going to be using.

Found the ESL so much easier than the dialogic c library we were using.

The docs for the esl are easy to understand, the thing I couldnt get
my head around initially was the dialing out, with the dialogic you
are in the middle when you dial, ie already on the channel, but with
freeswitch you are kind of the third party when you dial, the channel
being created by the dialing and handing it off to be worked on. We
make the call via an inbound event socket and hand it off to an
outbound event socket application via the dialplan.

On Wed, May 26, 2010 at 10:08 AM, Phone <phone.bytes at gmail.com> wrote:
> Thanks for the info. ?What language did you use?
>
> Bob Coleman wrote:
>> Have just recently completed a project to convert an old windows
>> dialogic application(vb6) to FreeSWITCH, would reccommend using the
>> ESL, was able to map the old dialogic calls to the ESL calls pretty
>> easily. We used a mixture of inbound and outbound sockets, as we have
>> people dialing us, not just dialing out etc.
>>
>> With the dialogic you open a port and make the call and handle the
>> dtmf, with freeswitch you create a socket connection to FreeSWITCH to
>> dial the number and then hand it off to an extension for processing
>> the dtmf(that is one approach any way)
>>
>> Bob
>>
>> On Wed, May 26, 2010 at 6:45 AM, Michael Collins <msc at freeswitch.org>
wrote:
>>
>>> On Tue, May 25, 2010 at 11:38 AM, Jan Berger <jan.berger at video24.no>
wrote:
>>>
>>>> Actually - before you get "to smart" - may I suggest that you start
>>>> writing
>>>> - or improving - the getting started sections of the doc. Address the
>>>> areas
>>>> where you struggle and let others benefit from your work.
>>>>
>>>> I have been through similar issues myself - FS is one of the easier
>>>> projects
>>>> to work with once you get under the hood, but you basically need to
evolve
>>>> to the level where you read the source code.
>>>>
>>>>
>>> And if you can wait 2+ months for "the book" then that should help as
well.
>>> :D
>>> -MC
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

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



From bobc at devassert.com  Tue May 25 20:04:31 2010
From: bobc at devassert.com (Bob Coleman)
Date: Wed, 26 May 2010 15:04:31 +1200
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
In-Reply-To: <D311BDCB06324D0DB77898FC6FF4B0EB@dell9400>
References: <490aedfdd7c8ac680bbe0d7f68c786de@mail.gmail.com>
	<4BFBE6C2.5000703@gmail.com>
	<AANLkTilsHn9znajzF0Jj-QDu4je9QW7CeymuBYE0SyKa@mail.gmail.com>
	<4BFBEF7C.9020107@gmail.com>
	<9250A9FF05DB4B879DEF6ECE8E3FFAC1@dell9400>
	<AANLkTilU2OwgH0Qk_Qoc3R7ErPO_Z_VRhePusetkRkeW@mail.gmail.com>
	<AANLkTikXWvEB_C3jqOWLHaVcns9-U5Hok-5MqgE_uzzZ@mail.gmail.com>
	<4BFC4A64.5090208@gmail.com>
	<AANLkTinFtVXa7CPou-ShPYIXHq0AktokhptGSEFKBkge@mail.gmail.com>
	<D311BDCB06324D0DB77898FC6FF4B0EB@dell9400>
Message-ID: <AANLkTilAMe1lRKK4R6Nmr29FzfI-pP3wRa7Foabmd8Dp@mail.gmail.com>

Ah sorry, I started with the esl to get an understanding then wrote my
own socket library(was actually very easy to do), when I mean docs I
mean the event socket docs. I still think of it as the esl, my
mistake.

http://wiki.freeswitch.org/wiki/Event_Socket

I started with a codeplex project, that had been abandoned, and then
once I understood the structure of the event socket language, was able
to rewrite it to better handle what we were doing.

I also married it up to an old gotdotnet asterisk fast agi project,
once again abandoned, to allow for the use of asterisk as well, but in
the end freeswitch won because we could use just one platform.

I am busy writing a small sample app at the moment to demonstrate a
problem I am trying to solve. Can release that code once sorted. Will
be in a week or so. Am intending it as a quick way of testing event
sockets, and trying various commands etc. before commiting to coding
something.

Bob

On Wed, May 26, 2010 at 2:26 PM, Jan Berger <jan.berger at video24.no> wrote:
> Hi,
>
> Do you have some sample code you could share + what docs did you look at?
>
> I would like to write and test some C# using ESL for my own work.
>
> Jan
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bob
> Coleman
> Sent: 26. mai 2010 04:07
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Questions on Building an application for
> FreeSWITCH
>
> We used c# as the rest of our systems are windows based. The language
> doesnt matter too much, as long as you know where you are headed, what
> performance you require, and what platform you are going to be using.
>
> Found the ESL so much easier than the dialogic c library we were using.
>
> The docs for the esl are easy to understand, the thing I couldnt get
> my head around initially was the dialing out, with the dialogic you
> are in the middle when you dial, ie already on the channel, but with
> freeswitch you are kind of the third party when you dial, the channel
> being created by the dialing and handing it off to be worked on. We
> make the call via an inbound event socket and hand it off to an
> outbound event socket application via the dialplan.
>
> On Wed, May 26, 2010 at 10:08 AM, Phone <phone.bytes at gmail.com> wrote:
>> Thanks for the info. ?What language did you use?
>>
>> Bob Coleman wrote:
>>> Have just recently completed a project to convert an old windows
>>> dialogic application(vb6) to FreeSWITCH, would reccommend using the
>>> ESL, was able to map the old dialogic calls to the ESL calls pretty
>>> easily. We used a mixture of inbound and outbound sockets, as we have
>>> people dialing us, not just dialing out etc.
>>>
>>> With the dialogic you open a port and make the call and handle the
>>> dtmf, with freeswitch you create a socket connection to FreeSWITCH to
>>> dial the number and then hand it off to an extension for processing
>>> the dtmf(that is one approach any way)
>>>
>>> Bob
>>>
>>> On Wed, May 26, 2010 at 6:45 AM, Michael Collins <msc at freeswitch.org>
> wrote:
>>>
>>>> On Tue, May 25, 2010 at 11:38 AM, Jan Berger <jan.berger at video24.no>
> wrote:
>>>>
>>>>> Actually - before you get "to smart" - may I suggest that you start
>>>>> writing
>>>>> - or improving - the getting started sections of the doc. Address the
>>>>> areas
>>>>> where you struggle and let others benefit from your work.
>>>>>
>>>>> I have been through similar issues myself - FS is one of the easier
>>>>> projects
>>>>> to work with once you get under the hood, but you basically need to
> evolve
>>>>> to the level where you read the source code.
>>>>>
>>>>>
>>>> And if you can wait 2+ months for "the book" then that should help as
> well.
>>>> :D
>>>> -MC
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From ranjtech at gmail.com  Tue May 25 21:34:02 2010
From: ranjtech at gmail.com (RR)
Date: Wed, 26 May 2010 00:34:02 -0400
Subject: [Freeswitch-users] Direct inward dialling
In-Reply-To: <AANLkTimMkeVcO7mqzCVA95PE_XSGoGdHfC2A7HJ5i0_R@mail.gmail.com>
References: <491B305B.4060307@gmx.net>
	<072A175C-9004-4C14-90EC-9A93A8453787@gmail.com>
	<AANLkTinxYIG6S30XMB5GUVuWsXlyhG_slqc_e0PQy0wM@mail.gmail.com>
	<AANLkTinjUuCyrK6eVSalV6fUVNFM-j_zlvrzRxq3lJZQ@mail.gmail.com>
	<AANLkTik0JvNDXax6Ay99hrcgzuiItY0f4LkCsaRjYS8a@mail.gmail.com>
	<AANLkTinYufK3FWtbM95088fOOOWWzOyMpU6ho0VXiOEy@mail.gmail.com>
	<AANLkTinLTWiKrvgPMiiEZArMxMQFCgtJNXoHO_BBlHRN@mail.gmail.com>
	<AANLkTilq_gLLPa-VaRlaiKofqNVlpvm1zgSiL53UTf7z@mail.gmail.com>
	<AANLkTikiraunYdhjcCUmKLmgNAyQ-Sp6r4EpohHuEbZn@mail.gmail.com>
	<AANLkTimMkeVcO7mqzCVA95PE_XSGoGdHfC2A7HJ5i0_R@mail.gmail.com>
Message-ID: <AANLkTim06fDEoeHe_hVVtfMDUeS2saTi-JyKiJQFB9ZC@mail.gmail.com>

Michael,

Thank you SO SO much for the help. Your regex work perfectly as desired. I
had tried what you suggested earlier but I think I might've made a mistake
somewhere because I wasn't getting the right results so I resorted to doing
the "|" between the prefixes to strip them out thinking maybe FS works by
going if it begins + OR +1 OR 011 then remove them but I guess it doesn't as
when "\" appears it uses / matches against only the first one of the those
as opposed to all of those.

Thanks again and sorry for wasting your time ;)

Cheers
RR

On Tue, May 25, 2010 at 9:19 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Tue, May 25, 2010 at 5:44 PM, RR <ranjtech at gmail.com> wrote:
>
>> Michael, haha, yeah they indeed are. That's why I'm routing based on $2,
>> but I still see the 1 and/or the 011 going through to the "bridge"
>> application. Why??
>>
> Because your regex is wrong. :) It took me a while to figure it out. I'm
> surprised it worked at all. All the stuff you have inside the first set of
> parens is not behaving the way you think it should be. If I read your
> intentions correctly you're trying to strip off leading:
> +
> OR
> +1
> OR
> 1
>
> In the first regex. Correct? If ANI is NANPA-ish then try this in your
> first regex:
> ^\+?1?([2-9]\d+).*$
>
> That should strip off leading + and/or 1 and capture just the 10-digit
> phone number in $1. (Be sure to use $1 and not $2, unless you had your heart
> set on using $2 in which case wrap the first part of the regex in parens)
>
> The other regex is also tricky. I assume you are trying to strip off the
> same as above as well as 011? Try this:
> ^\+?1?(011)?([2-9]\d+).*$
>
> Again, if the phone number in question is NANPA then $2 should contain just
> the 10 digits you want. Play around with that and let us know what happens.
> Also, don't forget what I said about using regex from the fs_cli. You can
> test all this stuff yourself. :)
>
> -MC
>
>
>
>
>>
>> On Tue, May 25, 2010 at 8:34 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>>
>>>
>>> On Tue, May 25, 2010 at 5:27 PM, RR <ranjtech at gmail.com> wrote:
>>>
>>>> Ok, so I take that back. This seems to only work when the dialplan has a
>>>> specific ANI and DNIS / destination_number / sip_to_user defined. If this is
>>>> more general
>>>>
>>>> like
>>>>
>>>> <include>
>>>>   <extension name="public_did">
>>>>     <condition field="ani" expression="^(\+?|\+1?|1?)(\d+).*$"
>>>> break="never">
>>>>         <action application="set" data="effective_caller_id_number=$2"/>
>>>>         <action application="set" data="effective_caller_id_name=$2"/>
>>>>     </condition>
>>>>     <condition field="${sip_to_user}"
>>>> expression="^(\+1?|\+|1?|011?)(\d+).*$" break="never">
>>>>       <action application="set" data="continue_on_fail=false"/>
>>>>       <action application="set" data="hangup_after_bridge=true"/>
>>>>       <action application="set" data="domain_name=$${domain}"/>
>>>>       <action application="set" data="bypass_media=true"/>
>>>>       <action application="limit_hash" data="in cc_blades 4200
>>>> !USER_BUSY"/>
>>>>       <action application="bridge"
>>>> data="{sip_invite_domain=${sip_from_host}}sofia/gateway/${distributor(cc_blades)}/$2"/>
>>>>     </condition>
>>>>   </extension>
>>>> </include>
>>>>
>>>> then even though the expression/conditions seem to match, none of the
>>>> digits are being stripped off. Shouldn't this be stripping off digits??
>>>>
>>>> Here's the debug output:
>>>>
>>>> Dialplan: sofia/external/16469NNNNNN Regex (PASS) [public_did]
>>>> ani(16469NNNNNN;phone-context=+1) =~ /^(\+?|\+1?|1?)(\d+).*$/ break=never
>>>> Dialplan: sofia/external/16469NNNNNN Action
>>>> set(effective_caller_id_number=16469NNNNNN)
>>>> Dialplan: sofia/external/16469NNNNNN Action
>>>> set(effective_caller_id_name=16469NNNNNN)
>>>> Dialplan: sofia/external/16469NNNNNN Regex (PASS) [public_did]
>>>> ${sip_to_user}(011390NNNNNNNNNN;phone-context=+39) =~
>>>> /^(\+1?|\+|1?|011?)(\d+).*$/ break=never
>>>> Dialplan: sofia/external/16469NNNNNN Action set(continue_on_fail=false)
>>>> Dialplan: sofia/external/16469NNNNNN Action
>>>> set(hangup_after_bridge=true)
>>>> Dialplan: sofia/external/16469NNNNNN Action
>>>> set(domain_name=208.72.186.166)
>>>> Dialplan: sofia/external/16469NNNNNN Action set(bypass_media=true)
>>>> Dialplan: sofia/external/16469NNNNNN Action limit_hash(in cc_blades 4200
>>>> !USER_BUSY)
>>>> Dialplan: sofia/external/16469NNNNNN Action
>>>> bridge({sip_invite_domain=${sip_from_host}}sofia/gateway/${distributor(cc_blades)}/011390NNNNNNNNNN)
>>>>
>>>> why're the '1' in the ANI and '011' in the DNIS/sip_to_user being
>>>> stripped off???
>>>>
>>>
>>> Regex 101 :)
>>>
>>> The 1 or the 011 are in $1
>>> -MC
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue May 25 22:40:30 2010
From: msc at freeswitch.org (Michael S Collins)
Date: Tue, 25 May 2010 22:40:30 -0700
Subject: [Freeswitch-users] Direct inward dialling
In-Reply-To: <AANLkTim06fDEoeHe_hVVtfMDUeS2saTi-JyKiJQFB9ZC@mail.gmail.com>
References: <491B305B.4060307@gmx.net>
	<072A175C-9004-4C14-90EC-9A93A8453787@gmail.com>
	<AANLkTinxYIG6S30XMB5GUVuWsXlyhG_slqc_e0PQy0wM@mail.gmail.com>
	<AANLkTinjUuCyrK6eVSalV6fUVNFM-j_zlvrzRxq3lJZQ@mail.gmail.com>
	<AANLkTik0JvNDXax6Ay99hrcgzuiItY0f4LkCsaRjYS8a@mail.gmail.com>
	<AANLkTinYufK3FWtbM95088fOOOWWzOyMpU6ho0VXiOEy@mail.gmail.com>
	<AANLkTinLTWiKrvgPMiiEZArMxMQFCgtJNXoHO_BBlHRN@mail.gmail.com>
	<AANLkTilq_gLLPa-VaRlaiKofqNVlpvm1zgSiL53UTf7z@mail.gmail.com>
	<AANLkTikiraunYdhjcCUmKLmgNAyQ-Sp6r4EpohHuEbZn@mail.gmail.com>
	<AANLkTimMkeVcO7mqzCVA95PE_XSGoGdHfC2A7HJ5i0_R@mail.gmail.com>
	<AANLkTim06fDEoeHe_hVVtfMDUeS2saTi-JyKiJQFB9ZC@mail.gmail.com>
Message-ID: <35FA7A4D-4937-4EA1-8FA4-B124129C92DD@freeswitch.org>

It's all good. Now you have to pay it forward. :)
-MC

Sent from my iPhone

On May 25, 2010, at 9:34 PM, RR <ranjtech at gmail.com> wrote:

> Michael,
>
> Thank you SO SO much for the help. Your regex work perfectly as  
> desired. I had tried what you suggested earlier but I think I  
> might've made a mistake somewhere because I wasn't getting the right  
> results so I resorted to doing the "|" between the prefixes to strip  
> them out thinking maybe FS works by going if it begins + OR +1 OR  
> 011 then remove them but I guess it doesn't as when "\" appears it  
> uses / matches against only the first one of the those as opposed to  
> all of those.
>
> Thanks again and sorry for wasting your time ;)
>
> Cheers
> RR
>
> On Tue, May 25, 2010 at 9:19 PM, Michael Collins  
> <msc at freeswitch.org> wrote:
>
>
> On Tue, May 25, 2010 at 5:44 PM, RR <ranjtech at gmail.com> wrote:
> Michael, haha, yeah they indeed are. That's why I'm routing based on  
> $2, but I still see the 1 and/or the 011 going through to the  
> "bridge" application. Why??
> Because your regex is wrong. :) It took me a while to figure it out.  
> I'm surprised it worked at all. All the stuff you have inside the  
> first set of parens is not behaving the way you think it should be.  
> If I read your intentions correctly you're trying to strip off  
> leading:
> +
> OR
> +1
> OR
> 1
>
> In the first regex. Correct? If ANI is NANPA-ish then try this in  
> your first regex:
> ^\+?1?([2-9]\d+).*$
>
> That should strip off leading + and/or 1 and capture just the 10- 
> digit phone number in $1. (Be sure to use $1 and not $2, unless you  
> had your heart set on using $2 in which case wrap the first part of  
> the regex in parens)
>
> The other regex is also tricky. I assume you are trying to strip off  
> the same as above as well as 011? Try this:
> ^\+?1?(011)?([2-9]\d+).*$
>
> Again, if the phone number in question is NANPA then $2 should  
> contain just the 10 digits you want. Play around with that and let  
> us know what happens. Also, don't forget what I said about using  
> regex from the fs_cli. You can test all this stuff yourself. :)
>
> -MC
>
>
>
>
> On Tue, May 25, 2010 at 8:34 PM, Michael Collins  
> <msc at freeswitch.org> wrote:
>
>
> On Tue, May 25, 2010 at 5:27 PM, RR <ranjtech at gmail.com> wrote:
> Ok, so I take that back. This seems to only work when the dialplan  
> has a specific ANI and DNIS / destination_number / sip_to_user  
> defined. If this is more general
>
> like
>
> <include>
>   <extension name="public_did">
>     <condition field="ani" expression="^(\+?|\+1?|1?)(\d+).*$"  
> break="never">
>         <action application="set" data="effective_caller_id_number= 
> $2"/>
>         <action application="set" data="effective_caller_id_name=$2"/>
>     </condition>
>     <condition field="${sip_to_user}" expression="^(\+1?|\+|1?|011?) 
> (\d+).*$" break="never">
>       <action application="set" data="continue_on_fail=false"/>
>       <action application="set" data="hangup_after_bridge=true"/>
>       <action application="set" data="domain_name=$${domain}"/>
>       <action application="set" data="bypass_media=true"/>
>       <action application="limit_hash" data="in cc_blades 4200 ! 
> USER_BUSY"/>
>       <action application="bridge" data="{sip_invite_domain=$ 
> {sip_from_host}}sofia/gateway/${distributor(cc_blades)}/$2"/>
>     </condition>
>   </extension>
> </include>
>
> then even though the expression/conditions seem to match, none of  
> the digits are being stripped off. Shouldn't this be stripping off  
> digits??
>
> Here's the debug output:
>
> Dialplan: sofia/external/16469NNNNNN Regex (PASS) [public_did] ani 
> (16469NNNNNN;phone-context=+1) =~ /^(\+?|\+1?|1?)(\d+).*$/ break=never
> Dialplan: sofia/external/16469NNNNNN Action set 
> (effective_caller_id_number=16469NNNNNN)
> Dialplan: sofia/external/16469NNNNNN Action set 
> (effective_caller_id_name=16469NNNNNN)
> Dialplan: sofia/external/16469NNNNNN Regex (PASS) [public_did] $ 
> {sip_to_user}(011390NNNNNNNNNN;phone-context=+39) =~ /^(\+1?|\+|1?| 
> 011?)(\d+).*$/ break=never
> Dialplan: sofia/external/16469NNNNNN Action set 
> (continue_on_fail=false)
> Dialplan: sofia/external/16469NNNNNN Action set 
> (hangup_after_bridge=true)
> Dialplan: sofia/external/16469NNNNNN Action set(domain_name=208.72.186.166 
> )
> Dialplan: sofia/external/16469NNNNNN Action set(bypass_media=true)
> Dialplan: sofia/external/16469NNNNNN Action limit_hash(in cc_blades  
> 4200 !USER_BUSY)
> Dialplan: sofia/external/16469NNNNNN Action bridge 
> ({sip_invite_domain=${sip_from_host}}sofia/gateway/${distributor 
> (cc_blades)}/011390NNNNNNNNNN)
>
> why're the '1' in the ANI and '011' in the DNIS/sip_to_user being  
> stripped off???
>
> Regex 101 :)
>
> The 1 or the 011 are in $1
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
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From michaelt at voxcore.voxtelecom.co.za  Wed May 26 01:19:01 2010
From: michaelt at voxcore.voxtelecom.co.za (Michael Toop)
Date: Wed, 26 May 2010 10:19:01 +0200
Subject: [Freeswitch-users] Simple IAX2 setup - help with converting
	from asterisk to freeswitch
In-Reply-To: <4D6421C5-9336-40D2-B54C-F773B2E6BA0E@jerris.com>
References: <91093d24f2166d220acf2ab19115f95d.squirrel@www.webcontracts.co.uk>
	<E1E3F6EF-1A1E-4891-B371-7A3BF0988E95@jerris.com>
	<d2976552e9ba0ce1ed095b7cef22e1ad.squirrel@www.webcontracts.co.uk>
	<4D6421C5-9336-40D2-B54C-F773B2E6BA0E@jerris.com>
Message-ID: <AANLkTimML1jF63o40HegxlE0reH-GEsOM4R410GtGcn3@mail.gmail.com>

Hi,

 Can someone assist me with how one would achieve what Michael Jerris
suggested: "IAX offers less overhead on rtp, but the same thing can be
accomplished with rtp using packetization..."   ...?

 I have done tons of reading on cRTP etc & I must be missing something as
there is no obvious stock standard Linux way to achieve what IAX give you?

Kind Regards,

Michael


On Fri, Feb 5, 2010 at 5:54 PM, Michael Jerris <mike at jerris.com> wrote:

> I find the secure and efficiency claims on IAX to be pretty much a farce.
>  IAX offers less overhead on rtp, but the same thing can be accomplished
> with rtp using packetization, as for security, I don't see any credible
> claim on that.  IAX also forces the program to sort out a ton of audio for
> different users going to 1 socket, something that a network stack is quite
> good at when using different ports, but is a lot more work where we are
> getting the packets.  As for default passwords and users, of course I
> wouldn't use those in production, those are for you to see how the pieces
> work together out of the box.  I wouldn't however quickly scrap the entire
> default config, just read through them and think about what you need and do
> not.  The extension ranges you use is totally at your discretion.
>
> Mike
>
>
> On Feb 5, 2010, at 8:53 AM, Matthew Law wrote:
>
> > Why is that? - a lot of web pages I have read claim that IAX is more
> > secure and efficient.  I have no problem with using SIP whatsoever and it
> > certainly appears to be ubiquitous.  I am a complete newcomer to VoIP and
> > I am trying to do this as securely as possible since I want to run
> > freeswitch on a Xen VPS which will be visible on the internet.
> >
> > Anyway, since my first question, I have worked my way through the wiki,
> > read a lot of example configs and made some sense of the docs.  I now
> have
> > a very basic config working (with SIP) on a local vmware image using the
> > 'quick and dirty' Makefile method.  I removed all of the example configs
> > from the xml files (those default extensions and passwords scared me) and
> > added just one for extension 1000, plus my dialplan and inbound/outbound
> > settings.
> >
> > One question: is there any reason not to use a smaller extension number
> > range, like 200-210, for example?
> >
> > Now to figure out how to get time based roaming working?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From codecomplete at free.fr  Wed May 26 01:32:41 2010
From: codecomplete at free.fr (GillesToo)
Date: Wed, 26 May 2010 01:32:41 -0700 (PDT)
Subject: [Freeswitch-users] make current -> "Failed to add directory
	'libs/openzap'"
In-Reply-To: <AANLkTimrQz07P-M1fLLMx5cd9ErVMJFjq_gS0sv9BfKP@mail.gmail.com>
References: <1274816996591-5100197.post@n2.nabble.com>
	<AANLkTimLE3l8hYCbyNYZPk3qcRpoCtXcVH5c3lunxaJv@mail.gmail.com>
	<1274822027357-5100501.post@n2.nabble.com>
	<AANLkTimrQz07P-M1fLLMx5cd9ErVMJFjq_gS0sv9BfKP@mail.gmail.com>
Message-ID: <1274862761273-5102426.post@n2.nabble.com>


Thanks Milena.

For those interested, here's how to install Git:
http://wiki.freeswitch.org/wiki/Git_Install

Note: The latest Git source code can be found here:
http://www.kernel.org/pub/software/scm/git/
(Looks like git-core is some old software, and you only need git.*.bz2)

And here's how to download the latest FS source through Git:
http://wiki.freeswitch.org/wiki/Installation_Guide#Recommended_Download


-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/make-current-Failed-to-add-directory-libs-openzap-tp5100197p5102426.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steveayre at gmail.com  Wed May 26 02:47:40 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 26 May 2010 10:47:40 +0100
Subject: [Freeswitch-users] G723 codec
In-Reply-To: <1055665185-1274834591-cardhu_decombobulator_blackberry.rim.net-1455303023-@bda057.bisx.prodap.on.blackberry>
References: <1055665185-1274834591-cardhu_decombobulator_blackberry.rim.net-1455303023-@bda057.bisx.prodap.on.blackberry>
Message-ID: <AANLkTil15RefXVZob01gESAItMfAq37YuiaU4y4vCTAx@mail.gmail.com>

There is G.723.1 passthrough-only support (mod_g723_1).

There's no G.723 support (which is an obsolete codec completely different to
G.723.1 and superceded by G.726 which is fully supported).

-Steve


On 26 May 2010 01:43, Budi wibowo <bwibowo at gmail.com> wrote:

> Hi, is it g723 in fs? Or just pass through?
> Other idea regarding codec issue is put transcoding box outside fs box.
>  You will also have flexibility for conncted client.
> Any client any codec can connect
>
> Br
> Budi
> ------Original Message------
> From: Steve Underwood
> Sender: freeswitch-users-bounces at lists.freeswitch.org
> To: freeswitch-users at lists.freeswitch.org
> ReplyTo: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] G723 codec
> Sent: May 26, 2010 01:07
>
> On 05/26/2010 01:42 AM, Madovsky wrote:
> > Hi all,
> > is there a chance that in future there will be a mod_g723 transcoder ?
> >
> I assume you mean G.723.1. It is very difficult to licence the patents
> related to G.723.1 unless you are a huge user. G.729 has a well defined
> scheme for licencing the relevant patents on an incremental basis,
> starting at a modest numbers of channels. G.723.1 is quite problematic.
> For that reason it is unlikely that G.723.1 will be available as a host
> software transcoder module any time soon. There are, however, hardware
> card options which can provide this codec.
>
> Steve
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> http://www.freeswitch.org
>
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From steveayre at gmail.com  Wed May 26 03:38:52 2010
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 26 May 2010 11:38:52 +0100
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
In-Reply-To: <AANLkTikTRItehHsdKYi6NUHbIWKrHtaTlT6fJQ8MHV2J@mail.gmail.com>
References: <0b27b9ae92e848c6edfcb0b6b20d255f@mail.gmail.com>
	<AANLkTiloS2j7QKkitjSmhECZXANvXnX8NH3fZiBL1qnm@mail.gmail.com>
	<4BEB2B7D.4080406@gmail.com>
	<AANLkTil9T_6yFvKA8FuoicbvpeZOIkI0XRguUkgO25hL@mail.gmail.com>
	<AANLkTikLpby-J0X6BvvutW7XxpgWylqZM3jU6SrREwBp@mail.gmail.com>
	<4BEB3E83.3040502@gmail.com>
	<AANLkTimNvBYVRBcbK3T9z5y80hLcs9BCp4uvyRZIuCWp@mail.gmail.com>
	<AANLkTik0JtsIKBLs-6ZJ2PxVq-esBBJe4dGiAiNLKhyG@mail.gmail.com>
	<AANLkTin_GcxWwk7otWFDzlA95XnoYSiVgvyky1mHLzGG@mail.gmail.com>
	<AANLkTikTRItehHsdKYi6NUHbIWKrHtaTlT6fJQ8MHV2J@mail.gmail.com>
Message-ID: <AANLkTikedHlOXNOXRcMoDZI7nuN6aDOJoZLcPVXs21Yn@mail.gmail.com>

But if that ESL server fails, your entire system goes down. Having N ESL
servers means one can fail and the other ESL server allows the cluster to
continue running until the failed server is replaced/repaired.

-Steve


On 13 May 2010 21:16, Phillip Jones <pjintheusa at gmail.com> wrote:

> What would the advantage of an N to N architecture be though? An ESL server
> controlling several FS instances has a view of everything that is going on.
> All calls/conferences etc. A real advantage. That is lost when two ESL
> servers are run in parallel. State information could be in a DB cluster -
> but why not have the in-process app access this directly, cutting out the
> middle man?
>
> How does using ESL make it more scalable and more available with fewer
> components? I am sure you are correct - I just don't see it.
>
>
> On Thu, May 13, 2010 at 3:15 PM, Kristian Kielhofner <kris at kriskinc.com>wrote:
>
>> ESL connecting to a socket actually proves to be more scalable and
>> more available with fewer components.  Why not have N servers running
>> your socket app with N servers running FreeSWITCH?
>>
>> OpenSIPS introduces its own issues with failover and I've yet to see
>> DNS SRV be the reliability/scalability solution it's made out to be.
>>
>> On Thu, May 13, 2010 at 3:01 PM, Phillip Jones <pjintheusa at gmail.com>
>> wrote:
>> > And just a general thought of using ESL vs an in process solution like
>> > mod_managed, or LUA.
>> >
>> > My understanding is, that using a separate server/process does
>> potentially
>> > give you another point of failure and, if you use a single ESL server
>> > application to control several FS boxes, potentially a single point of
>> > failure. It is fairly easy to build a scalable and reliable FS cluster
>> and
>> > using DNS SRV and OpenSIPS in order to avoid any single points of
>> failure.
>> > Having independent FS boxes that pull data, but can fail with little
>> impact
>> > seems attractive to me,
>>
>> --
>> Kristian Kielhofner
>> http://www.astlinux.org
>> http://blog.krisk.org
>> http://www.star2star.com
>> http://www.submityoursip.com
>> http://www.voalte.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From codecomplete at free.fr  Wed May 26 03:50:17 2010
From: codecomplete at free.fr (GillesToo)
Date: Wed, 26 May 2010 03:50:17 -0700 (PDT)
Subject: [Freeswitch-users] [init.d] Waiting until FS is actually ready?
Message-ID: <1274871017473-5102826.post@n2.nabble.com>


Hello,

On a CentOS host, I use the RedHat init script to start up Freeswitch at
boot time:

----
http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Running_FreeSWITCH

cp build/freeswitch.init.redhat /etc/init.d/freeswitch
chmod 755 /etc/init.d/freeswitch
----

However, I also use this script to start the server semi-manually when doing
tests, but the script returns "OK" right away, even though Freeswitch is
still in the process of starting up (eg. NAT, PMP, etc.) so I have to wait
about 20 seconds before running fs_cli so I can connect to it without an
error.

What should I add/change to the init script so that it only returns "OK"
once FS is really up and running?

Thank you.

-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/init-d-Waiting-until-FS-is-actually-ready-tp5102826p5102826.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From davis.erwin at gmail.com  Wed May 26 04:23:29 2010
From: davis.erwin at gmail.com (Erwin Davis)
Date: Wed, 26 May 2010 07:23:29 -0400
Subject: [Freeswitch-users] call to the phone registering with xml_curl
	failed
Message-ID: <AANLkTil875dYNTRzbmAnuwE9USqv44jX6u-NjjIgISnz@mail.gmail.com>

Hi, I have two phone (x-lite), one (ext 1018) registering with xml_curl and
the other (ext 1001) registering locally.
I was able to make call from ext 1018 to ext 1001. But the call from ext
1001 to ext 1018 failed. I cut and pasted the console log in
http://pastebin.freeswitch.org/13044. Any clue? Thanks,

e
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From jan.berger at video24.no  Wed May 26 04:50:21 2010
From: jan.berger at video24.no (Jan Berger)
Date: Wed, 26 May 2010 13:50:21 +0200
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
In-Reply-To: <AANLkTikedHlOXNOXRcMoDZI7nuN6aDOJoZLcPVXs21Yn@mail.gmail.com>
References: <0b27b9ae92e848c6edfcb0b6b20d255f@mail.gmail.com>
	<AANLkTiloS2j7QKkitjSmhECZXANvXnX8NH3fZiBL1qnm@mail.gmail.com>
	<4BEB2B7D.4080406@gmail.com>
	<AANLkTil9T_6yFvKA8FuoicbvpeZOIkI0XRguUkgO25hL@mail.gmail.com>
	<AANLkTikLpby-J0X6BvvutW7XxpgWylqZM3jU6SrREwBp@mail.gmail.com>
	<4BEB3E83.3040502@gmail.com>
	<AANLkTimNvBYVRBcbK3T9z5y80hLcs9BCp4uvyRZIuCWp@mail.gmail.com>
	<AANLkTik0JtsIKBLs-6ZJ2PxVq-esBBJe4dGiAiNLKhyG@mail.gmail.com>
	<AANLkTin_GcxWwk7otWFDzlA95XnoYSiVgvyky1mHLzGG@mail.gmail.com>
	<AANLkTikTRItehHsdKYi6NUHbIWKrHtaTlT6fJQ8MHV2J@mail.gmail.com>
	<AANLkTikedHlOXNOXRcMoDZI7nuN6aDOJoZLcPVXs21Yn@mail.gmail.com>
Message-ID: <960AB94CA4D3483590AD59913A16B60D@dell9400>

N+N pools can be used for many things in telecom, but not for all. 

 

An IVR can as an example be a pool - rather than FS running the IVR server
side you connect to a ESL resource and run heavy work client side, maybe
through a pool manager. On this level you can even provide failover schemes
and fall-back schemes. Speech servers are another excellent candidate for
resource pools because speech requires a bit of CPU and it's very nice to be
able to scale up speech capacity by just throwing in more hardware.

 

The challenge comes when you want to do things like ACD on a N+N system. As
you agent's now might be connected to different FS boxes you need a
centralised instance to manage - in which case you can use N2 or simply use
a database that provide redundancy. 

 

Jan

 

  _____  

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven
Ayre
Sent: 26. mai 2010 12:39
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Questions on Building an application for
FreeSWITCH

 

But if that ESL server fails, your entire system goes down. Having N ESL
servers means one can fail and the other ESL server allows the cluster to
continue running until the failed server is replaced/repaired.

-Steve



On 13 May 2010 21:16, Phillip Jones <pjintheusa at gmail.com> wrote:

What would the advantage of an N to N architecture be though? An ESL server
controlling several FS instances has a view of everything that is going on.
All calls/conferences etc. A real advantage. That is lost when two ESL
servers are run in parallel. State information could be in a DB cluster -
but why not have the in-process app access this directly, cutting out the
middle man? 

How does using ESL make it more scalable and more available with fewer
components? I am sure you are correct - I just don't see it.

 

On Thu, May 13, 2010 at 3:15 PM, Kristian Kielhofner <kris at kriskinc.com>
wrote:

ESL connecting to a socket actually proves to be more scalable and
more available with fewer components.  Why not have N servers running
your socket app with N servers running FreeSWITCH?

OpenSIPS introduces its own issues with failover and I've yet to see
DNS SRV be the reliability/scalability solution it's made out to be.


On Thu, May 13, 2010 at 3:01 PM, Phillip Jones <pjintheusa at gmail.com> wrote:
> And just a general thought of using ESL vs an in process solution like
> mod_managed, or LUA.
>
> My understanding is, that using a separate server/process does potentially
> give you another point of failure and, if you use a single ESL server
> application to control several FS boxes, potentially a single point of
> failure. It is fairly easy to build a scalable and reliable FS cluster and
> using DNS SRV and OpenSIPS in order to avoid any single points of failure.
> Having independent FS boxes that pull data, but can fail with little
impact
> seems attractive to me,

--
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com


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

 


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

 

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From freeswitch-list at puzzled.xs4all.nl  Wed May 26 05:10:32 2010
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Wed, 26 May 2010 14:10:32 +0200
Subject: [Freeswitch-users] [init.d] Waiting until FS is actually ready?
In-Reply-To: <1274871017473-5102826.post@n2.nabble.com>
References: <1274871017473-5102826.post@n2.nabble.com>
Message-ID: <4BFD0FB8.3060701@puzzled.xs4all.nl>

On 05/26/2010 12:50 PM, GillesToo wrote:
[snip]
>
> What should I add/change to the init script so that it only returns "OK"
> once FS is really up and running?

I don't think there is a solution for this as (iirc) the "OK" is based 
on the resultcode from starting "daemon" and not from starting 
freeswitch itself.

Regards,
Patrick


From codecomplete at free.fr  Wed May 26 06:37:08 2010
From: codecomplete at free.fr (GillesToo)
Date: Wed, 26 May 2010 06:37:08 -0700 (PDT)
Subject: [Freeswitch-users] [Out of curiosity] Why not use a voice modem for
	FXO?
Message-ID: <1274881028458-5103502.post@n2.nabble.com>


Hello

In case there are system developers out there with good experience writing
modem softare, considering that...
1. even a cheap OpenVox PCI card starts at $90 for just one FXO port
2. even entry-level PC's theses days have ample CPU power especially when
just used a basic server with Linux
3. we've been using so-called softmodems (ie. controller-less modems that
offload processing to the computer's CPU) for over a decade and they sell
for about $10 as OEM
4. and finally, Digium/Asterisk's Zaptel driver was precisely meant to run
on cheap voice cards

... I was wondering why no one has gone forth and written a driver for this
kind of hardware, which would be great for SOHO users to handle just one
landline. Is it because the hardware is not standard enough, and writing
this kind of driver takes a lot of work?

Thank you.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Out-of-curiosity-Why-not-use-a-voice-modem-for-FXO-tp5103502p5103502.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From brian at freeswitch.org  Wed May 26 06:42:00 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 26 May 2010 08:42:00 -0500
Subject: [Freeswitch-users] [Out of curiosity] Why not use a voice modem
	for FXO?
In-Reply-To: <1274881028458-5103502.post@n2.nabble.com>
References: <1274881028458-5103502.post@n2.nabble.com>
Message-ID: <6596AAE7-B489-4D4A-B661-EDE2CD351DC3@freeswitch.org>

What do you mean?  Anything that is zaptel compatible will work with OpenZAP/FreeTDM.  Even the cheap OpenVox single port FXO's.

/b

On May 26, 2010, at 8:37 AM, GillesToo wrote:

> ... I was wondering why no one has gone forth and written a driver for this
> kind of hardware, which would be great for SOHO users to handle just one
> landline. Is it because the hardware is not standard enough, and writing
> this kind of driver takes a lot of work?



From davis.erwin at gmail.com  Wed May 26 06:55:31 2010
From: davis.erwin at gmail.com (Erwin Davis)
Date: Wed, 26 May 2010 09:55:31 -0400
Subject: [Freeswitch-users] call to the phone registering with xml_curl
	failed
In-Reply-To: <AANLkTil875dYNTRzbmAnuwE9USqv44jX6u-NjjIgISnz@mail.gmail.com>
References: <AANLkTil875dYNTRzbmAnuwE9USqv44jX6u-NjjIgISnz@mail.gmail.com>
Message-ID: <AANLkTinjcDQJhsLDWBa-8JZ5AbUIhh3UO2G6NllAELCD@mail.gmail.com>

I fixed it with dialplan change.

On Wed, May 26, 2010 at 7:23 AM, Erwin Davis <davis.erwin at gmail.com> wrote:

> Hi, I have two phone (x-lite), one (ext 1018) registering with xml_curl and
> the other (ext 1001) registering locally.
> I was able to make call from ext 1018 to ext 1001. But the call from ext
> 1001 to ext 1018 failed. I cut and pasted the console log in
> http://pastebin.freeswitch.org/13044. Any clue? Thanks,
>
> e
>
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From codecomplete at free.fr  Wed May 26 06:59:07 2010
From: codecomplete at free.fr (GillesToo)
Date: Wed, 26 May 2010 06:59:07 -0700 (PDT)
Subject: [Freeswitch-users] [Out of curiosity] Why not use a voice modem
	for FXO?
In-Reply-To: <6596AAE7-B489-4D4A-B661-EDE2CD351DC3@freeswitch.org>
References: <1274881028458-5103502.post@n2.nabble.com>
	<6596AAE7-B489-4D4A-B661-EDE2CD351DC3@freeswitch.org>
Message-ID: <1274882347926-5103647.post@n2.nabble.com>


What do you mean?  Anything that is zaptel compatible will work with
OpenZAP/FreeTDM.  Even the cheap OpenVox single port FXO's. 


Sorry: I mean that Zaptel doesn't work on softmodems, just Digium and
Digium-compatible telephony cards. I don't see why softmodems couldn't
handle the load of a single landline.

For instance, "U.S. Robotics USR5670 56Kbps PCI Bus (Plug & Play) Fax modem"
sells for $17:

http://www.newegg.com/Product/Product.aspx?Item=N82E16825104001
www.newegg.com/Product/Product.aspx?Item=N82E16825104001 

Or maybe the hardware out there is just too diverse that writing a driver
for this is a PITA?
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Out-of-curiosity-Why-not-use-a-voice-modem-for-FXO-tp5103502p5103647.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From codecomplete at free.fr  Wed May 26 07:05:01 2010
From: codecomplete at free.fr (GillesToo)
Date: Wed, 26 May 2010 07:05:01 -0700 (PDT)
Subject: [Freeswitch-users] [init.d] Waiting until FS is actually ready?
In-Reply-To: <4BFD0FB8.3060701@puzzled.xs4all.nl>
References: <1274871017473-5102826.post@n2.nabble.com>
	<4BFD0FB8.3060701@puzzled.xs4all.nl>
Message-ID: <1274882701743-5103683.post@n2.nabble.com>


Too bad, but I can live with that. Thanks.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/init-d-Waiting-until-FS-is-actually-ready-tp5102826p5103683.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From jingwei.yang at gmail.com  Wed May 26 07:19:45 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Wed, 26 May 2010 22:19:45 +0800
Subject: [Freeswitch-users]  FreeSwitchCoreLib compilation error
Message-ID: <AANLkTilVIRh-PDJQDuDns_Os2jgTBxYaoo81k8UYfK18@mail.gmail.com>

Hello,

I just updated the source code to the latest revision and hit an error when
compiling FSComm. It happened in switch_cpp.cpp, at line 1139.

SWITCH_DECLARE(int) globalSetVariable(const char *var, const char *val,
const char *val2)
{
    if (zstr(val)) val = NULL;
    if (zstr(val2)) val2 = NULL;

    if (val2) {
        switch_core_set_var_conditional(var, val, val2);
    } else {
        switch_core_set_variable(var, val);
    }
}

Is it supposed to return a value? Error message: error C4716:
'globalSetVariable' : must return a value

Regards,
-Jingwei
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From peter.olsson at visionutveckling.se  Wed May 26 07:27:44 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 26 May 2010 16:27:44 +0200
Subject: [Freeswitch-users] FreeSwitchCoreLib compilation error
In-Reply-To: <AANLkTilVIRh-PDJQDuDns_Os2jgTBxYaoo81k8UYfK18@mail.gmail.com>
References: <AANLkTilVIRh-PDJQDuDns_Os2jgTBxYaoo81k8UYfK18@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C557DE1E087@cooper>

I just reported this on jira (with a patch). FSBUILD-278.

/Peter


Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Jingwei Yang
Skickat: den 26 maj 2010 16:20
Till: freeswitch-users
?mne: [Freeswitch-users] FreeSwitchCoreLib compilation error

Hello,

I just updated the source code to the latest revision and hit an error when compiling FSComm. It happened in switch_cpp.cpp, at line 1139.
SWITCH_DECLARE(int) globalSetVariable(const char *var, const char *val, const char *val2)
{
    if (zstr(val)) val = NULL;
    if (zstr(val2)) val2 = NULL;

    if (val2) {
        switch_core_set_var_conditional(var, val, val2);
    } else {
        switch_core_set_variable(var, val);
    }
}

Is it supposed to return a value? Error message: error C4716: 'globalSetVariable' : must return a value

Regards,
-Jingwei
!DSPAM:4bfd2fad32931800758995!
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From brian at freeswitch.org  Wed May 26 07:35:53 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 26 May 2010 09:35:53 -0500
Subject: [Freeswitch-users] FreeSwitchCoreLib compilation error
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C557DE1E087@cooper>
References: <AANLkTilVIRh-PDJQDuDns_Os2jgTBxYaoo81k8UYfK18@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C557DE1E087@cooper>
Message-ID: <D85E0127-853B-49E4-9EA9-9FD84E4767AE@freeswitch.org>

Committed.  Thanks.

/b

On May 26, 2010, at 9:27 AM, Peter Olsson wrote:

> I just reported this on jira (with a patch). FSBUILD-278.
>  
> /Peter
>  

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From anthony.minessale at gmail.com  Wed May 26 07:39:54 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 26 May 2010 09:39:54 -0500
Subject: [Freeswitch-users] [init.d] Waiting until FS is actually ready?
In-Reply-To: <1274882701743-5103683.post@n2.nabble.com>
References: <1274871017473-5102826.post@n2.nabble.com>
	<4BFD0FB8.3060701@puzzled.xs4all.nl>
	<1274882701743-5103683.post@n2.nabble.com>
Message-ID: <AANLkTimMpiFEah5vshcxOjYiXkqQWxsQR4EYkZ_ekWUt@mail.gmail.com>

You could code in a while loop into the script to run

fs_cli -x "status"

until it works, but it would require mod_event_socket



On Wed, May 26, 2010 at 9:05 AM, GillesToo <codecomplete at free.fr> wrote:

>
> Too bad, but I can live with that. Thanks.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/init-d-Waiting-until-FS-is-actually-ready-tp5102826p5103683.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
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
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From anthony.minessale at gmail.com  Wed May 26 07:56:26 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 26 May 2010 09:56:26 -0500
Subject: [Freeswitch-users] Simple IAX2 setup - help with converting
	from asterisk to freeswitch
In-Reply-To: <AANLkTimML1jF63o40HegxlE0reH-GEsOM4R410GtGcn3@mail.gmail.com>
References: <91093d24f2166d220acf2ab19115f95d.squirrel@www.webcontracts.co.uk>
	<E1E3F6EF-1A1E-4891-B371-7A3BF0988E95@jerris.com>
	<d2976552e9ba0ce1ed095b7cef22e1ad.squirrel@www.webcontracts.co.uk>
	<4D6421C5-9336-40D2-B54C-F773B2E6BA0E@jerris.com>
	<AANLkTimML1jF63o40HegxlE0reH-GEsOM4R410GtGcn3@mail.gmail.com>
Message-ID: <AANLkTinwP2RJtukpGx1ISPdPBtwETlbkh2v0Hm_bnkZR@mail.gmail.com>

Well what IAX gives us on FreeSWITCH is random segfaults because the
protocol changed in a way
that made the IAX stack library we were using a deathtrap.  The authors of
IAX decided to change the protocol
so that it was not backwards compatible with their own previous work and due
to lack of resources we decided that removing it was the best way
to achieve stability.

IAX offers some nat relief with the 1 port strategy at the expense of the
stack implementation in user space.
Frankly the authors of this protocol failed to supply a robust reference
implementation of the protocol in a standalone license free package.  There
was one small library designed specifically for a standalone client (that's
the one we used) and as I explained, in the end, it failed us.

It then boils down to the business question: Do I want to chase after
marginal savings on my configuration and packet sizes at the expense of much
fewer calls and a serious risk of catastrophic failure caused by a poor
implementation?

SIP has its flaws as well, we don't have enough time to get into that, but
on FreeSWITCH it can scale much higher than IAX even when it was working
better (before they changed it).

You said you are new to VoIP (which is surprising based on your email addr)
so from your perspective what is it that you are looking for?

rtp packetization refers to sending more audio at once to reduce the
overhead of IP headers.
in FreeSWITCH you can configure your codecs to PCMU at 60i for instance to
bundle 3 typical packets into 1

BTW, No matter what you use, you will have little luck on a virtual machine
as your first experience.
I recommend you get a real box (multi-core 64bit) to play with.


On Wed, May 26, 2010 at 3:19 AM, Michael Toop <
michaelt at voxcore.voxtelecom.co.za> wrote:

> Hi,
>
>  Can someone assist me with how one would achieve what Michael Jerris
> suggested: "IAX offers less overhead on rtp, but the same thing can be
> accomplished with rtp using packetization..."   ...?
>
>  I have done tons of reading on cRTP etc & I must be missing something as
> there is no obvious stock standard Linux way to achieve what IAX give you?
>
> Kind Regards,
>
> Michael
>
>
> On Fri, Feb 5, 2010 at 5:54 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> I find the secure and efficiency claims on IAX to be pretty much a farce.
>>  IAX offers less overhead on rtp, but the same thing can be accomplished
>> with rtp using packetization, as for security, I don't see any credible
>> claim on that.  IAX also forces the program to sort out a ton of audio for
>> different users going to 1 socket, something that a network stack is quite
>> good at when using different ports, but is a lot more work where we are
>> getting the packets.  As for default passwords and users, of course I
>> wouldn't use those in production, those are for you to see how the pieces
>> work together out of the box.  I wouldn't however quickly scrap the entire
>> default config, just read through them and think about what you need and do
>> not.  The extension ranges you use is totally at your discretion.
>>
>> Mike
>>
>>
>> On Feb 5, 2010, at 8:53 AM, Matthew Law wrote:
>>
>> > Why is that? - a lot of web pages I have read claim that IAX is more
>> > secure and efficient.  I have no problem with using SIP whatsoever and
>> it
>> > certainly appears to be ubiquitous.  I am a complete newcomer to VoIP
>> and
>> > I am trying to do this as securely as possible since I want to run
>> > freeswitch on a Xen VPS which will be visible on the internet.
>> >
>> > Anyway, since my first question, I have worked my way through the wiki,
>> > read a lot of example configs and made some sense of the docs.  I now
>> have
>> > a very basic config working (with SIP) on a local vmware image using the
>> > 'quick and dirty' Makefile method.  I removed all of the example configs
>> > from the xml files (those default extensions and passwords scared me)
>> and
>> > added just one for extension 1000, plus my dialplan and inbound/outbound
>> > settings.
>> >
>> > One question: is there any reason not to use a smaller extension number
>> > range, like 200-210, for example?
>> >
>> > Now to figure out how to get time based roaming working?
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
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
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From anthony.minessale at gmail.com  Wed May 26 07:59:51 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 26 May 2010 09:59:51 -0500
Subject: [Freeswitch-users] [Out of curiosity] Why not use a voice modem
	for FXO?
In-Reply-To: <1274882347926-5103647.post@n2.nabble.com>
References: <1274881028458-5103502.post@n2.nabble.com>
	<6596AAE7-B489-4D4A-B661-EDE2CD351DC3@freeswitch.org>
	<1274882347926-5103647.post@n2.nabble.com>
Message-ID: <AANLkTikeqyqds4FVor8qBZ9PZgrz9dIPATPAbr2N8TCQ@mail.gmail.com>

Even if you get it working the reward is not worth the effort.
You would experience a lot of echo and other problems and the time you spent
working on it would
be better spent doing something more meaningful and using the money you
received to buy a real TDM card or
better still, a FXO->SIP ATA.

On Wed, May 26, 2010 at 8:59 AM, GillesToo <codecomplete at free.fr> wrote:

>
> What do you mean?  Anything that is zaptel compatible will work with
> OpenZAP/FreeTDM.  Even the cheap OpenVox single port FXO's.
>
>
> Sorry: I mean that Zaptel doesn't work on softmodems, just Digium and
> Digium-compatible telephony cards. I don't see why softmodems couldn't
> handle the load of a single landline.
>
> For instance, "U.S. Robotics USR5670 56Kbps PCI Bus (Plug & Play) Fax
> modem"
> sells for $17:
>
> http://www.newegg.com/Product/Product.aspx?Item=N82E16825104001
> www.newegg.com/Product/Product.aspx?Item=N82E16825104001
>
> Or maybe the hardware out there is just too diverse that writing a driver
> for this is a PITA?
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Out-of-curiosity-Why-not-use-a-voice-modem-for-FXO-tp5103502p5103647.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
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
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From jingwei.yang at gmail.com  Wed May 26 08:26:27 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Wed, 26 May 2010 23:26:27 +0800
Subject: [Freeswitch-users] FreeSwitchCoreLib compilation error
In-Reply-To: <D85E0127-853B-49E4-9EA9-9FD84E4767AE@freeswitch.org>
References: <AANLkTilVIRh-PDJQDuDns_Os2jgTBxYaoo81k8UYfK18@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C557DE1E087@cooper>
	<D85E0127-853B-49E4-9EA9-9FD84E4767AE@freeswitch.org>
Message-ID: <AANLkTikND1kWy8xaQKusQ7BoQsD_Pptb2kh1buBqkSti@mail.gmail.com>

Thanks Peter and Brian,

By the way, do you guys know the following link errors? It occurred when I
was building FSComm in VS2008. The steps I followed are from this link:
http://wiki.freeswitch.org/wiki/FSComm#Windows. I have installed QT and set
the QTDIR environment variable. Besides, all the necessary dlls have been
copied to the debug folder. Please help hint where might go wrong.

Thanks,
-Jingwei

10>Linking...

10>fshost.obj : error LNK2019: unresolved external symbol "public:
__thiscall Channel::Channel(class QString)" (??0Channel@@QAE at VQString@@@Z)
referenced in function "private: void __thiscall
FSHost::eventChannelCreate(class QSharedPointer<struct switch_event>,class
QString)" (?eventChannelCreate at FSHost@@AAEXV?$QSharedPointer at Uswitch_event
@@@@VQString@@@Z)

10>mainwindow.obj : error LNK2019: unresolved external symbol "public:
__thiscall StateDebugDialog::StateDebugDialog(class QWidget *)"
(??0StateDebugDialog@@QAE at PAVQWidget@@@Z) referenced in function "private:
void __thiscall MainWindow::debugEventsTriggered(void)"
(?debugEventsTriggered at MainWindow@@AAEXXZ)

10>mainwindow.obj : error LNK2019: unresolved external symbol "public:
__thiscall ConsoleWindow::ConsoleWindow(class QWidget *)" (??0ConsoleWindow@
@QAE at PAVQWidget@@@Z) referenced in function "private: void __thiscall
MainWindow::debugConsoleTriggered(void)" (?debugConsoleTriggered at MainWindow
@@AAEXXZ)

10>prefdialog.obj : error LNK2019: unresolved external symbol "public:
__thiscall CodecWidget::CodecWidget(class QWidget *)" (??0CodecWidget@
@QAE at PAVQWidget@@@Z) referenced in function "public: void __thiscall
Ui_PrefDialog::setupUi(class QDialog *)" (?setupUi at Ui_PrefDialog
@@QAEXPAVQDialog@@@Z)

10>prefsofia.obj : error LNK2019: unresolved external symbol "public: void
__thiscall CodecWidget::setCodecString(class QString)"
(?setCodecString at CodecWidget@@QAEXVQString@@@Z) referenced in function
"public: void __thiscall PrefSofia::readConfig(void)" (?readConfig at PrefSofia
@@QAEXXZ)

10>prefsofia.obj : error LNK2019: unresolved external symbol "public: class
QString __thiscall CodecWidget::getCodecString(void)"
(?getCodecString at CodecWidget@@QAE?AVQString@@XZ) referenced in function
"public: void __thiscall PrefSofia::writeConfig(void)"
(?writeConfig at PrefSofia@@QAEXXZ)

10>debug\FSComm.exe : fatal error LNK1120: 6 unresolved externals



On Wed, May 26, 2010 at 10:35 PM, Brian West <brian at freeswitch.org> wrote:

> Committed.  Thanks.
>
> /b
>
> On May 26, 2010, at 9:27 AM, Peter Olsson wrote:
>
> I just reported this on jira (with a patch). FSBUILD-278.
>
> /Peter
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dswardstrom at remotelink.com  Wed May 26 08:29:42 2010
From: dswardstrom at remotelink.com (David Swardstrom)
Date: Wed, 26 May 2010 10:29:42 -0500 (CDT)
Subject: [Freeswitch-users] libtool version mismatch
In-Reply-To: <56730863.59.1274887671091.JavaMail.root@srvr12.remotelinkml.com>
Message-ID: <801457633.61.1274887782484.JavaMail.root@srvr12.remotelinkml.com>

Recently my Ubuntu system did an update.
I am now getting the following when a do a make current.

<quote>
WARNING mod_xml_odbc is not a valid FreeSWITCH module dir, skipping it.....

making clean mod_yaml
Making clean in .
Making clean in .
 rm -f freeswitch fs_cli fs_ivrd
rm -f "./so_locations"
rm -f /usr/local/freeswitch//usr/local/freeswitch/mod/*.so
Making uninstall in .
 ( cd '/usr/local/freeswitch/bin' && rm -f freeswitch fs_cli fs_ivrd )
 ( cd '/usr/local/freeswitch/bin' && rm -f gentls_cert fsxs )
 /bin/bash /usr/src/freeswitch/quiet_libtool   --mode=uninstall rm -f '/usr/local/freeswitch/lib/libfreeswitch.la'
quiet_libtool: Version mismatch error.  This is libtool 2.2.6 Debian-2.2.6a-4, but the
quiet_libtool: definition of this LT_INIT comes from libtool 2.2.6b.
quiet_libtool: You should recreate aclocal.m4 with macros from libtool 2.2.6 Debian-2.2.6a-4
quiet_libtool: and run autoconf again.
make[1]: *** [uninstall-libLTLIBRARIES] Error 63
make: *** [uninstall-recursive] Error 1
<quote/>

I have tried various things including re-executing:
<quote>
./bootstrap.sh
./configure
<quote/>

Can I resolve this by deleting a file (or files) in /usr/src/freeswitch
and then letting the system recreate or reload them?

Regards,
Paul David Swardstrom



From brian at freeswitch.org  Wed May 26 08:32:44 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 26 May 2010 10:32:44 -0500
Subject: [Freeswitch-users] libtool version mismatch
In-Reply-To: <801457633.61.1274887782484.JavaMail.root@srvr12.remotelinkml.com>
References: <801457633.61.1274887782484.JavaMail.root@srvr12.remotelinkml.com>
Message-ID: <564963C6-9C70-4224-B853-3AEF436E01F4@freeswitch.org>

Wipe your tree and rebootstrap.

/b

On May 26, 2010, at 10:29 AM, David Swardstrom wrote:

> Recently my Ubuntu system did an update.
> I am now getting the following when a do a make current.



From mike at yes.net.ua  Wed May 26 08:41:54 2010
From: mike at yes.net.ua (Mike Tkachuk)
Date: Wed, 26 May 2010 18:41:54 +0300
Subject: [Freeswitch-users] [Out of curiosity] Why not use a voice modem
	for FXO?
In-Reply-To: <1274881028458-5103502.post@n2.nabble.com>
References: <1274881028458-5103502.post@n2.nabble.com>
Message-ID: <78238665.20100526184154@yes.net.ua>

Hello GillesToo,

 Asterisk "X100P" is just Intel IA92 WinModem.
 This is the so-called "clone" X100P, you can buy it on ebay for 15$
 I used this last time like 5 years ago :)
 Main  problems  you'll  face  is echo and detecting line states (when
 remote party hangup or pick up a call)

Wednesday, May 26, 2010 4:37:08 PM, you wrote:


G> Hello

G> In case there are system developers out there with good experience writing
G> modem softare, considering that...
G> 1. even a cheap OpenVox PCI card starts at $90 for just one FXO port
G> 2. even entry-level PC's theses days have ample CPU power especially when
G> just used a basic server with Linux
G> 3. we've been using so-called softmodems (ie. controller-less modems that
G> offload processing to the computer's CPU) for over a decade and they sell
G> for about $10 as OEM
G> 4. and finally, Digium/Asterisk's Zaptel driver was precisely meant to run
G> on cheap voice cards

G> ... I was wondering why no one has gone forth and written a driver for this
G> kind of hardware, which would be great for SOHO users to handle just one
G> landline. Is it because the hardware is not standard enough, and writing
G> this kind of driver takes a lot of work?

G> Thank you.



--
Mike Tkachuk



From steveu at coppice.org  Wed May 26 08:53:46 2010
From: steveu at coppice.org (Steve Underwood)
Date: Wed, 26 May 2010 23:53:46 +0800
Subject: [Freeswitch-users] [Out of curiosity] Why not use a voice modem
 for	FXO?
In-Reply-To: <1274881028458-5103502.post@n2.nabble.com>
References: <1274881028458-5103502.post@n2.nabble.com>
Message-ID: <4BFD440A.6020902@coppice.org>

On 05/26/2010 09:37 PM, GillesToo wrote:
> Hello
>
> In case there are system developers out there with good experience writing
> modem softare, considering that...
> 1. even a cheap OpenVox PCI card starts at $90 for just one FXO port
> 2. even entry-level PC's theses days have ample CPU power especially when
> just used a basic server with Linux
> 3. we've been using so-called softmodems (ie. controller-less modems that
> offload processing to the computer's CPU) for over a decade and they sell
> for about $10 as OEM
> 4. and finally, Digium/Asterisk's Zaptel driver was precisely meant to run
> on cheap voice cards
>
> ... I was wondering why no one has gone forth and written a driver for this
> kind of hardware, which would be great for SOHO users to handle just one
> landline. Is it because the hardware is not standard enough, and writing
> this kind of driver takes a lot of work?
>
> Thank you.
>    
Despite a lot of hot air spouted to the contrary, very few people have 
produced open source telephony code. None of those have so far been 
motivated to produce the relevant driver, so today the only one 
available is the Digium's own intel 536 driver in the zaptel/dahdi 
distribution. If you would like to produce drivers for other common 
chipsets, many people would be delighted. They just won't contribute.

Steve



From phone.bytes at gmail.com  Wed May 26 08:59:30 2010
From: phone.bytes at gmail.com (Phone)
Date: Wed, 26 May 2010 09:59:30 -0600
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
In-Reply-To: <AANLkTilAMe1lRKK4R6Nmr29FzfI-pP3wRa7Foabmd8Dp@mail.gmail.com>
References: <490aedfdd7c8ac680bbe0d7f68c786de@mail.gmail.com>	<4BFBE6C2.5000703@gmail.com>	<AANLkTilsHn9znajzF0Jj-QDu4je9QW7CeymuBYE0SyKa@mail.gmail.com>	<4BFBEF7C.9020107@gmail.com>	<9250A9FF05DB4B879DEF6ECE8E3FFAC1@dell9400>	<AANLkTilU2OwgH0Qk_Qoc3R7ErPO_Z_VRhePusetkRkeW@mail.gmail.com>	<AANLkTikXWvEB_C3jqOWLHaVcns9-U5Hok-5MqgE_uzzZ@mail.gmail.com>	<4BFC4A64.5090208@gmail.com>	<AANLkTinFtVXa7CPou-ShPYIXHq0AktokhptGSEFKBkge@mail.gmail.com>	<D311BDCB06324D0DB77898FC6FF4B0EB@dell9400>
	<AANLkTilAMe1lRKK4R6Nmr29FzfI-pP3wRa7Foabmd8Dp@mail.gmail.com>
Message-ID: <4BFD4562.5030804@gmail.com>

Thanks to all for the most helpful feedback.  Sharing your approaches 
and experiences are a big help.  I look forward to the upcoming code 
samples.

I was coming from a windows/dialogic environment where I used a library 
that allowed me to work on a little higher level.  For example, I had a 
call to "play a file" that took a parameter of whether or not to allow a 
dtmf to interrupt.  There was also a call to "ReadDtmfs" that took 
parameters to specify the number of Dtmf's to read, how long to wait for 
them, and what terminating character to use.  I guess that you could 
write some scripts or compiled code with these same types of functions 
to simplify some of these routine tasks with reusable code?

Also, the library handled the threading and scheduling with the OS.  I 
am still unclear on handling the events.  I guess you have a big loop 
reading events and then acting on them using the uuid to determine which 
call it is and how to deal with the next step of the call?  Any feedback 
on this part of the project?

Again, Thanks!

Bob Coleman wrote:
> Ah sorry, I started with the esl to get an understanding then wrote my
> own socket library(was actually very easy to do), when I mean docs I
> mean the event socket docs. I still think of it as the esl, my
> mistake.
>
> http://wiki.freeswitch.org/wiki/Event_Socket
>
> I started with a codeplex project, that had been abandoned, and then
> once I understood the structure of the event socket language, was able
> to rewrite it to better handle what we were doing.
>
> I also married it up to an old gotdotnet asterisk fast agi project,
> once again abandoned, to allow for the use of asterisk as well, but in
> the end freeswitch won because we could use just one platform.
>
> I am busy writing a small sample app at the moment to demonstrate a
> problem I am trying to solve. Can release that code once sorted. Will
> be in a week or so. Am intending it as a quick way of testing event
> sockets, and trying various commands etc. before commiting to coding
> something.
>
> Bob
>
> On Wed, May 26, 2010 at 2:26 PM, Jan Berger <jan.berger at video24.no> wrote:
>   
>> Hi,
>>
>> Do you have some sample code you could share + what docs did you look at?
>>
>> I would like to write and test some C# using ESL for my own work.
>>
>> Jan
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bob
>> Coleman
>> Sent: 26. mai 2010 04:07
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Questions on Building an application for
>> FreeSWITCH
>>
>> We used c# as the rest of our systems are windows based. The language
>> doesnt matter too much, as long as you know where you are headed, what
>> performance you require, and what platform you are going to be using.
>>
>> Found the ESL so much easier than the dialogic c library we were using.
>>
>> The docs for the esl are easy to understand, the thing I couldnt get
>> my head around initially was the dialing out, with the dialogic you
>> are in the middle when you dial, ie already on the channel, but with
>> freeswitch you are kind of the third party when you dial, the channel
>> being created by the dialing and handing it off to be worked on. We
>> make the call via an inbound event socket and hand it off to an
>> outbound event socket application via the dialplan.
>>
>> On Wed, May 26, 2010 at 10:08 AM, Phone <phone.bytes at gmail.com> wrote:
>>     
>>> Thanks for the info.  What language did you use?
>>>
>>> Bob Coleman wrote:
>>>       
>>>> Have just recently completed a project to convert an old windows
>>>> dialogic application(vb6) to FreeSWITCH, would reccommend using the
>>>> ESL, was able to map the old dialogic calls to the ESL calls pretty
>>>> easily. We used a mixture of inbound and outbound sockets, as we have
>>>> people dialing us, not just dialing out etc.
>>>>
>>>> With the dialogic you open a port and make the call and handle the
>>>> dtmf, with freeswitch you create a socket connection to FreeSWITCH to
>>>> dial the number and then hand it off to an extension for processing
>>>> the dtmf(that is one approach any way)
>>>>
>>>> Bob
>>>>
>>>> On Wed, May 26, 2010 at 6:45 AM, Michael Collins <msc at freeswitch.org>
>>>>         
>> wrote:
>>     
>>>>> On Tue, May 25, 2010 at 11:38 AM, Jan Berger <jan.berger at video24.no>
>>>>>           
>> wrote:
>>     
>>>>>> Actually - before you get "to smart" - may I suggest that you start
>>>>>> writing
>>>>>> - or improving - the getting started sections of the doc. Address the
>>>>>> areas
>>>>>> where you struggle and let others benefit from your work.
>>>>>>
>>>>>> I have been through similar issues myself - FS is one of the easier
>>>>>> projects
>>>>>> to work with once you get under the hood, but you basically need to
>>>>>>             
>> evolve
>>     
>>>>>> to the level where you read the source code.
>>>>>>
>>>>>>
>>>>>>             
>>>>> And if you can wait 2+ months for "the book" then that should help as
>>>>>           
>> well.
>>     
>>>>> :D
>>>>> -MC
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>>           
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>         
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>       
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From gmaruzz at celliax.org  Wed May 26 09:32:48 2010
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 26 May 2010 18:32:48 +0200
Subject: [Freeswitch-users] [Out of curiosity] Why not use a voice modem
	for FXO?
In-Reply-To: <78238665.20100526184154@yes.net.ua>
References: <1274881028458-5103502.post@n2.nabble.com>
	<78238665.20100526184154@yes.net.ua>
Message-ID: <AANLkTin91ijwmPNy6wZC_G45MIfNTOYPMY-LAhaVvBj6@mail.gmail.com>

On Wed, May 26, 2010 at 5:41 PM, Mike Tkachuk <mike at yes.net.ua> wrote:
> Hello GillesToo,
>
> ?Asterisk "X100P" is just Intel IA92 WinModem.
> ?This is the so-called "clone" X100P, you can buy it on ebay for 15$
> ?I used this last time like 5 years ago :)
> ?Main ?problems ?you'll ?face ?is echo and detecting line states (when
> ?remote party hangup or pick up a call)

XP100 clones that you can buy cheap works decent if you use OSLEC line
echo cancellation (
http://www.rowetel.com/ucasterisk/oslec.html#install ).

Obviously, that will not give you the reliability, call flow analisys,
etc you can get from a "real" telephony card, or better an ATA, but
you can experiment with it and see if it can satisfy your needs (using
OSLEC for zaptel).


>
> Wednesday, May 26, 2010 4:37:08 PM, you wrote:
>
>
> G> Hello
>
> G> In case there are system developers out there with good experience writing
> G> modem softare, considering that...
> G> 1. even a cheap OpenVox PCI card starts at $90 for just one FXO port
> G> 2. even entry-level PC's theses days have ample CPU power especially when
> G> just used a basic server with Linux
> G> 3. we've been using so-called softmodems (ie. controller-less modems that
> G> offload processing to the computer's CPU) for over a decade and they sell
> G> for about $10 as OEM
> G> 4. and finally, Digium/Asterisk's Zaptel driver was precisely meant to run
> G> on cheap voice cards
>
> G> ... I was wondering why no one has gone forth and written a driver for this
> G> kind of hardware, which would be great for SOHO users to handle just one
> G> landline. Is it because the hardware is not standard enough, and writing
> G> this kind of driver takes a lot of work?
>
> G> Thank you.
>
>
>
> --
> Mike Tkachuk
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From jerry.richards at teotech.com  Wed May 26 09:41:30 2010
From: jerry.richards at teotech.com (Jerry Richards)
Date: Wed, 26 May 2010 09:41:30 -0700
Subject: [Freeswitch-users] Valet Parking Twice To Same Location
Message-ID: <26ED8669AF9949FD9ACA002C58834EF8@greyhawk.tonecommander.com>


If I inadvertantly park two calls to the same valet park location, the two
parked calls become bridged and are no longer available to be unparked.  Is
there a tag to prevent this or is that the intended behavior?

Thanks,
Jerry



From brian at freeswitch.org  Wed May 26 09:46:12 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 26 May 2010 11:46:12 -0500
Subject: [Freeswitch-users] Valet Parking Twice To Same Location
In-Reply-To: <26ED8669AF9949FD9ACA002C58834EF8@greyhawk.tonecommander.com>
References: <26ED8669AF9949FD9ACA002C58834EF8@greyhawk.tonecommander.com>
Message-ID: <8E6936F5-5E5C-43C0-A65C-0DAE0CC3726A@freeswitch.org>

That is the intended behavior.  You can do an extension to park people at random within a range... 

/b

On May 26, 2010, at 11:41 AM, Jerry Richards wrote:

> 
> If I inadvertantly park two calls to the same valet park location, the two
> parked calls become bridged and are no longer available to be unparked.  Is
> there a tag to prevent this or is that the intended behavior?
> 
> Thanks,
> Jerry



From msc at freeswitch.org  Wed May 26 09:59:37 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 26 May 2010 09:59:37 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call Starting Now!
Message-ID: <AANLkTind8C64Sb5EUXS70eBWHnof-8vSJztFunmTkodh@mail.gmail.com>

C'mon over:

http://wiki.freeswitch.org/wiki/FS_weekly_2010_05_26

-MC
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From dswardstrom at remotelink.com  Wed May 26 10:22:22 2010
From: dswardstrom at remotelink.com (David Swardstrom)
Date: Wed, 26 May 2010 12:22:22 -0500 (CDT)
Subject: [Freeswitch-users] libtool version mismatch
In-Reply-To: <1061904150.73.1274894251967.JavaMail.root@srvr12.remotelinkml.com>
Message-ID: <640313454.80.1274894541888.JavaMail.root@srvr12.remotelinkml.com>

I think I understand what "wipe your tree" means in a sense.
But I don't know what I need to do this.
I suspect that this means the I need to delete certain files in every
subdirectory but what files.

While I have used Unix based systems in the past, I never had to use make
because either some other build process was used or make was embedded
in other tools. For the last couple of years, I have been working
with Visual Studio and C++. 

I think that this "clean-up" everything related to the make and
remake would be something that could be useful to add to one of the
Wiki pages.

Regards,
Paul David Swardstrom



From msc at freeswitch.org  Wed May 26 11:39:44 2010
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 26 May 2010 11:39:44 -0700
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
In-Reply-To: <4BFD4562.5030804@gmail.com>
References: <490aedfdd7c8ac680bbe0d7f68c786de@mail.gmail.com>
	<4BFBEF7C.9020107@gmail.com>
	<9250A9FF05DB4B879DEF6ECE8E3FFAC1@dell9400>
	<AANLkTilU2OwgH0Qk_Qoc3R7ErPO_Z_VRhePusetkRkeW@mail.gmail.com>
	<AANLkTikXWvEB_C3jqOWLHaVcns9-U5Hok-5MqgE_uzzZ@mail.gmail.com>
	<4BFC4A64.5090208@gmail.com>
	<AANLkTinFtVXa7CPou-ShPYIXHq0AktokhptGSEFKBkge@mail.gmail.com>
	<D311BDCB06324D0DB77898FC6FF4B0EB@dell9400>
	<AANLkTilAMe1lRKK4R6Nmr29FzfI-pP3wRa7Foabmd8Dp@mail.gmail.com>
	<4BFD4562.5030804@gmail.com>
Message-ID: <AANLkTinIq6d_Y7XA6q8nlQtVEtr9aSqc_W3V93Fjn3kV@mail.gmail.com>

On Wed, May 26, 2010 at 8:59 AM, Phone <phone.bytes at gmail.com> wrote:

> Thanks to all for the most helpful feedback.  Sharing your approaches
> and experiences are a big help.  I look forward to the upcoming code
> samples.
>
No problem.

>
> I was coming from a windows/dialogic environment where I used a library
> that allowed me to work on a little higher level.  For example, I had a
> call to "play a file" that took a parameter of whether or not to allow a
> dtmf to interrupt.  There was also a call to "ReadDtmfs" that took
> parameters to specify the number of Dtmf's to read, how long to wait for
> them, and what terminating character to use.  I guess that you could
> write some scripts or compiled code with these same types of functions
> to simplify some of these routine tasks with reusable code?
>
FreeSWITCH has several tools to assist with this, most notably
playAndGetDigits
The dialplan app is doc'd here:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits
The IVR-ish way is doc'd here: http://wiki.freeswitch.org/wiki/IVR
The mod_lua docs have good info about using it from calling lua from the
dialplan: http://wiki.freeswitch.org/wiki/Mod_lua#session:playAndGetDigits


>
> Also, the library handled the threading and scheduling with the OS.  I
> am still unclear on handling the events.  I guess you have a big loop
> reading events and then acting on them using the uuid to determine which
> call it is and how to deal with the next step of the call?  Any feedback
> on this part of the project?
>
I think you might be suffering from information overload. :)
The event socket works in two methods: inbound and outbound. We have been
throwing around concepts that apply to both without really doing a lot of
qualifying.

In most cases we are talking about inbound event socket where your server
establishes a connection to FreeSWITCH. To see more about the outbound event
socket look at this:
http://wiki.freeswitch.org/wiki/Event_socket_outbound
Essentially you can create a program/script that sits and listens on a TCP
port for connections from FS. When your program gets a connection then it
can control the call leg via event socket commands. In the outbound event
socket paradigm your program sees *ONLY* the events related to your specific
call leg. When the call ends, so does your socket connection. You can use
outbound socket for call control if you want to do so, however it is by no
means required.

As far as inbound event socket goes, the official docs start here:
http://wiki.freeswitch.org/wiki/Event_socket

The classic example of an event socket application is one you may not
expect: fs_cli
If you REALLY want to see what the event socket looks like then fire up
fs_cli and do this:
/log 0
/events plain all

At that point your terminal is a raw event socket client. :) You can send
commands and see all the crazy stuff that comes across via events. Once you
get a feel for the events that come and go then you can think about how to
attack your problem.

In your specific question about using the uuid to figure out what to do with
the call... yes, you *could* do that if you wanted to. You could also have
the call controlled by the dialplan. For example, you originate a call and
drop it into the dialplan and let the dialplan handle it. You will still get
events about that channel, for examples when the caller changes states, or
even DTMFs, etc. (if you want to see them) but the call is being controlled
by the dialplan itself so that your program doesn't have to control it. You
could then have, as part of your event loop, a "show channels" (or "show
channels as xml") happen every x seconds and just parse it to see what's
happening on your calls. If you're a real stud programmer you could have
your script maintain a state machine for each call and use the incoming
events from the event socket to update your call states.

Are you sure you still want to do this? :)
-MC
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From max.clark at gmail.com  Wed May 26 11:35:32 2010
From: max.clark at gmail.com (Max Clark)
Date: Wed, 26 May 2010 11:35:32 -0700
Subject: [Freeswitch-users] Polycom Config Generator
Message-ID: <AANLkTimekEYLgSkRI9NmFwgjZ5TfHjYNNNrAhZSQ1C9J@mail.gmail.com>

Hello,

I'm looking for a Polycom configuration generator (tftp) to use with
FreeSWITCH. Does such a thing exist?

Thanks,
Max


From gkuri at ieee.org  Wed May 26 12:06:44 2010
From: gkuri at ieee.org (Gabriel Kuri)
Date: Wed, 26 May 2010 12:06:44 -0700
Subject: [Freeswitch-users] Polycom Config Generator
In-Reply-To: <AANLkTimekEYLgSkRI9NmFwgjZ5TfHjYNNNrAhZSQ1C9J@mail.gmail.com>
References: <AANLkTimekEYLgSkRI9NmFwgjZ5TfHjYNNNrAhZSQ1C9J@mail.gmail.com>
Message-ID: <AANLkTimlAx_AZX0eRu1IPb4Hr8Y6mo6auo7-UkB1ksQJ@mail.gmail.com>

Yes, it does exist, however I believe you need to get it through a Polycom
Partner. It's part of some downloadable SDK that only partners have access
to, AFAIK.

Cheers,
Gabe

On Wed, May 26, 2010 at 11:35 AM, Max Clark <max.clark at gmail.com> wrote:

> Hello,
>
> I'm looking for a Polycom configuration generator (tftp) to use with
> FreeSWITCH. Does such a thing exist?
>
> Thanks,
> Max
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Wed May 26 12:13:55 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 26 May 2010 14:13:55 -0500
Subject: [Freeswitch-users] Polycom Config Generator
In-Reply-To: <AANLkTimlAx_AZX0eRu1IPb4Hr8Y6mo6auo7-UkB1ksQJ@mail.gmail.com>
References: <AANLkTimekEYLgSkRI9NmFwgjZ5TfHjYNNNrAhZSQ1C9J@mail.gmail.com>
	<AANLkTimlAx_AZX0eRu1IPb4Hr8Y6mo6auo7-UkB1ksQJ@mail.gmail.com>
Message-ID: <98357B0E-BA21-4EEF-A9E5-F8D733E51FFD@freeswitch.org>

http://search.cpan.org/~zblair/Polycom-Config-File-0.03/

/b

On May 26, 2010, at 2:06 PM, Gabriel Kuri wrote:

> Yes, it does exist, however I believe you need to get it through a Polycom Partner. It's part of some downloadable SDK that only partners have access to, AFAIK.
> 
> Cheers,
> Gabe
> 
> On Wed, May 26, 2010 at 11:35 AM, Max Clark <max.clark at gmail.com> wrote:
> Hello,
> 
> I'm looking for a Polycom configuration generator (tftp) to use with
> FreeSWITCH. Does such a thing exist?
> 
> Thanks,
> Max
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From sid at eltc.ru  Wed May 26 03:49:16 2010
From: sid at eltc.ru (Sergey Scheglov)
Date: Wed, 26 May 2010 17:49:16 +0700
Subject: [Freeswitch-users] Don't work playback after bypass media mode.
Message-ID: <1427456515.20100526174916@eltc.ru>

Hi, Freeswitch-users.

My dialplan:
<extension name="Local_Extension">
    <condition field="destination_number" expression="^([1-9]\d{4})$">
        <action application="set" data="dialed_extension=$1"/>
        <action application="export" data="dialed_extension=$1"/>
        <action application="set" data="ringback=${ru-ring}"/>
        <action application="set" data="bypass_media=true"/>
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set" data="continue_on_fail=SUBSCRIBER_ABSENT,USER_NOT_REGISTERED"/>
        <action application="set" data="continue_on_fail=true"/>
        <action application="bridge" data="user/${dialed_extension}@${domain_name}"/>
        <action application="pre_answer"/>
        <action application="sleep" data="1000"/>
        <action application="playback" data="elight/${originate_disposition}.wav"/>
    </condition>
</extension>

If hangup cause is SUBSCRIBER_ABSENT or USER_NOT_REGISTERED, then
playback don't work.

Piece of log

2010-05-26 17:35:08.218253 [ERR] switch_ivr_originate.c:2493 Cannot create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT]
2010-05-26 17:35:08.218253 [DEBUG] switch_ivr_originate.c:3308 Originate Resulted in Error Cause: 20 [SUBSCRIBER_ABSENT]
2010-05-26 17:35:08.218253 [INFO] mod_dptools.c:2356 Originate Failed.  Cause: SUBSCRIBER_ABSENT
EXECUTE sofia/internal/39663 at 10.110.1.8:5060 pre_answer()
2010-05-26 17:35:08.219282 [INFO] mod_dptools.c:746 Sending early media
2010-05-26 17:35:08.219282 [NOTICE] mod_sofia.c:2032 Pre-Answer sofia/internal/39663 at 10.110.1.8:5060!
2010-05-26 17:35:08.219282 [DEBUG] switch_core_session.c:642 Send signal sofia/internal/39663 at 10.110.1.8:5060 [BREAK]
EXECUTE sofia/internal/39663 at 10.110.1.8:5060 sleep(1000)
2010-05-26 17:35:08.219282 [DEBUG] sofia.c:4195 Channel sofia/internal/39663 at 10.110.1.8:5060 skipping state [early][183]
2010-05-26 17:35:09.219252 [DEBUG] switch_core_session.c:642 Send signal sofia/internal/39663 at 10.110.1.8:5060 [BREAK]
2010-05-26 17:35:39.226271 [ERR] switch_core_io.c:121 sofia/internal/39663 at 10.110.1.8:5060 has no read codec.
2010-05-26 17:35:39.226271 [DEBUG] switch_core_session.c:1759 Application playback Requires media on channel sofia/internal/39663 at 10.110.1.8:5060!
EXECUTE sofia/internal/39663 at 10.110.1.8:5060 playback(elight/SUBSCRIBER_ABSENT.wav)
2010-05-26 17:35:39.226271 [NOTICE] switch_core_state_machine.c:185 sofia/internal/39663 at 10.110.1.8:5060 has executed the last dialplan instruction, hanging up.
2010-05-26 17:35:39.226271 [NOTICE] switch_core_state_machine.c:187 Hangup sofia/internal/39663 at 10.110.1.8:5060 [CS_EXECUTE] [NORMAL_CLEARING]

What am I doing wrong?

If set bypass_media_after_bridge instead bypass_media, then works
fine, BUT changing codec negotiation.

Thanks.
-- 
Regard
Sergey Scheglov





From emilbergg at gmail.com  Wed May 26 06:19:13 2010
From: emilbergg at gmail.com (Emil Berg)
Date: Wed, 26 May 2010 16:19:13 +0300
Subject: [Freeswitch-users] Disabling voice mail
Message-ID: <AANLkTil6J0PvUS4rZx2lBNbIhWIDELH5k9euUZ06J0Gq@mail.gmail.com>

Hello,

I am a newbie and I'd like to disable the voice mail.
All I want is that when calling to 1000-1010 users, the voice mail will not
answer in case that the user is not available. Instead, the server will send
a standard reject.
Is there any simple way to make it through configuration?

I was trying to change the dialplans, but it's very confusing and it didn't
work.

Thank you,
Emil.
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From brian at freeswitch.org  Wed May 26 12:51:34 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 26 May 2010 14:51:34 -0500
Subject: [Freeswitch-users] Disabling voice mail
In-Reply-To: <AANLkTil6J0PvUS4rZx2lBNbIhWIDELH5k9euUZ06J0Gq@mail.gmail.com>
References: <AANLkTil6J0PvUS4rZx2lBNbIhWIDELH5k9euUZ06J0Gq@mail.gmail.com>
Message-ID: <4C7FF608-66F3-4DAF-9C6F-F26C0D7FEB36@freeswitch.org>

Just remove it from the dialplan... 

/b

On May 26, 2010, at 8:19 AM, Emil Berg wrote:

> Hello,
> 
> I am a newbie and I'd like to disable the voice mail.
> All I want is that when calling to 1000-1010 users, the voice mail will not answer in case that the user is not available. Instead, the server will send a standard reject.
> Is there any simple way to make it through configuration?
> 
> I was trying to change the dialplans, but it's very confusing and it didn't work.
> 
> Thank you,
> Emil.



From brian at freeswitch.org  Wed May 26 12:52:05 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 26 May 2010 14:52:05 -0500
Subject: [Freeswitch-users] Don't work playback after bypass media mode.
In-Reply-To: <1427456515.20100526174916@eltc.ru>
References: <1427456515.20100526174916@eltc.ru>
Message-ID: <23EA73D6-FAF2-4067-84E0-6A2DDCB339E7@freeswitch.org>

What rev are you running?

/b

On May 26, 2010, at 5:49 AM, Sergey Scheglov wrote:

> 
> What am I doing wrong?
> 
> If set bypass_media_after_bridge instead bypass_media, then works
> fine, BUT changing codec negotiation.



From codecomplete at free.fr  Wed May 26 13:26:49 2010
From: codecomplete at free.fr (GillesToo)
Date: Wed, 26 May 2010 13:26:49 -0700 (PDT)
Subject: [Freeswitch-users] [Out of curiosity] Why not use a voice modem
	for FXO?
In-Reply-To: <AANLkTikeqyqds4FVor8qBZ9PZgrz9dIPATPAbr2N8TCQ@mail.gmail.com>
References: <1274881028458-5103502.post@n2.nabble.com>
	<6596AAE7-B489-4D4A-B661-EDE2CD351DC3@freeswitch.org>
	<1274882347926-5103647.post@n2.nabble.com>
	<AANLkTikeqyqds4FVor8qBZ9PZgrz9dIPATPAbr2N8TCQ@mail.gmail.com>
Message-ID: <1274905609840-5105604.post@n2.nabble.com>



Anthony Minessale wrote:
> Even if you get it working the reward is not worth the effort. You would
> experience a lot of echo and other problems

I read somewhere that the reason for those problems people often have with
X100P cards (such as echo or bad far-end disconnect supervision) is that
most of them have they use the Silicon Labs DAA chips Si3012/Si3035 which is
only good for use in countries that use the FCC standard; Countries that use
the CTR21 standard (Europe, and others) require the Si3014/Si3034 chips,
which support global line standards.


Anthony Minessale wrote:
> ... and the time you spent working on it would be better spent doing
> something more meaningful and using the money you received to buy a real
> TDM card or better still, a FXO->SIP ATA.

I thought TDM cards were a better solution to connect an IP PBX to a
landline. Why would you recommend an FXO-SIP ATA instead? And which
brand/model of ATA is recommended for use with Freeswitch?
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Out-of-curiosity-Why-not-use-a-voice-modem-for-FXO-tp5103502p5105604.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From leo.zibi at gmail.com  Wed May 26 13:37:05 2010
From: leo.zibi at gmail.com (leo.zibi at gmail.com)
Date: Wed, 26 May 2010 22:37:05 +0200
Subject: [Freeswitch-users] Disabling voice mail
In-Reply-To: <4C7FF608-66F3-4DAF-9C6F-F26C0D7FEB36@freeswitch.org>
References: <AANLkTil6J0PvUS4rZx2lBNbIhWIDELH5k9euUZ06J0Gq@mail.gmail.com>
	<4C7FF608-66F3-4DAF-9C6F-F26C0D7FEB36@freeswitch.org>
Message-ID: <4BFD8671.6020001@gmail.com>

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From codecomplete at free.fr  Wed May 26 13:40:59 2010
From: codecomplete at free.fr (GillesToo)
Date: Wed, 26 May 2010 13:40:59 -0700 (PDT)
Subject: [Freeswitch-users] [init.d] Waiting until FS is actually ready?
In-Reply-To: <AANLkTimMpiFEah5vshcxOjYiXkqQWxsQR4EYkZ_ekWUt@mail.gmail.com>
References: <1274871017473-5102826.post@n2.nabble.com>
	<4BFD0FB8.3060701@puzzled.xs4all.nl>
	<1274882701743-5103683.post@n2.nabble.com>
	<AANLkTimMpiFEah5vshcxOjYiXkqQWxsQR4EYkZ_ekWUt@mail.gmail.com>
Message-ID: <1274906459055-5105679.post@n2.nabble.com>



Anthony Minessale wrote:
> You could code in a while loop into the script to run
> 
> fs_cli -x "status"
> 
> until it works, but it would require mod_event_socket

Thanks for the tip.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/init-d-Waiting-until-FS-is-actually-ready-tp5102826p5105679.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Wed May 26 14:07:10 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 26 May 2010 16:07:10 -0500
Subject: [Freeswitch-users] [Out of curiosity] Why not use a voice modem
	for FXO?
In-Reply-To: <1274905609840-5105604.post@n2.nabble.com>
References: <1274881028458-5103502.post@n2.nabble.com>
	<6596AAE7-B489-4D4A-B661-EDE2CD351DC3@freeswitch.org>
	<1274882347926-5103647.post@n2.nabble.com>
	<AANLkTikeqyqds4FVor8qBZ9PZgrz9dIPATPAbr2N8TCQ@mail.gmail.com>
	<1274905609840-5105604.post@n2.nabble.com>
Message-ID: <AANLkTimvJNT2NPp63C-IeiII-BRTti_CNoWJJjKUmZVS@mail.gmail.com>

It all depends, FXO is never truly reliable with all the disconnect
supervision / hangup detection woes.
A FXO card works as well as it can but traditionally requires an
echo canceler if you want quality that
will not draw complaints from your best QA testers, your family for instance
who expect identical
quality to the PSTN.  The Sangoma cards for instance have this.

The good thing about an ATA is that it's self-contained and translates
everything into SIP for you when
you don't have room for hardware.

It's not worth cutting corners on telephone quality when, in the end, you
will replace it once you get enough
complaints.




On Wed, May 26, 2010 at 3:26 PM, GillesToo <codecomplete at free.fr> wrote:

>
>
> Anthony Minessale wrote:
> > Even if you get it working the reward is not worth the effort. You would
> > experience a lot of echo and other problems
>
> I read somewhere that the reason for those problems people often have with
> X100P cards (such as echo or bad far-end disconnect supervision) is that
> most of them have they use the Silicon Labs DAA chips Si3012/Si3035 which
> is
> only good for use in countries that use the FCC standard; Countries that
> use
> the CTR21 standard (Europe, and others) require the Si3014/Si3034 chips,
> which support global line standards.
>
>
> Anthony Minessale wrote:
> > ... and the time you spent working on it would be better spent doing
> > something more meaningful and using the money you received to buy a real
> > TDM card or better still, a FXO->SIP ATA.
>
> I thought TDM cards were a better solution to connect an IP PBX to a
> landline. Why would you recommend an FXO-SIP ATA instead? And which
> brand/model of ATA is recommended for use with Freeswitch?
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Out-of-curiosity-Why-not-use-a-voice-modem-for-FXO-tp5103502p5105604.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
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
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From sean at obscuradigital.com  Wed May 26 14:12:40 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Wed, 26 May 2010 14:12:40 -0700
Subject: [Freeswitch-users] Freetdm error
Message-ID: <C822DCD8.B1F5%sean@obscuradigital.com>

Hello all,

I?ve been hacking at this issue for the last 3 days and I can?t get beyond
this error when attempting to call out through my sangoma card.

  SIP/2.0 503 Service Unavailable
   Via: SIP/2.0/UDP
192.168.100.6:12026;branch=z9hG4bK-d8754z-77564c04563bac12-1---d8754z-;rport
=12026
   From: "Sean Holt"<sip:2028 at 172.16.15.10>;tag=25dacb3d
   To: <sip:5102079553 at 172.16.15.10>;tag=4St9j3N9yXr8g
   Call-ID: NWViZTI0ODljMmY1YTM5ZDRiZmQzNTUwYzgwNWVkMDQ.
   CSeq: 1 INVITE
   User-Agent: Configured by TCAPI
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: precondition, path, replaces
   Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Reason: Q.850;cause=34;text="NORMAL_CIRCUIT_CONGESTION"
   Content-Length: 0
   Remote-Party-ID: "5102079553"
<sip:5102079553 at 172.16.15.10>;party=calling;privacy=off;screen=no

Here?s my configuration:
I?m trying to call out from my internal voip phone through my sangoma card
b600.  My phone line is plugged into the FXO port.  Module (mod_freetdm) is
successfully loaded
Dialplan:
<extension name="main_number_53">
      <condition field="destination_number"
expression="^(\d{11})$|^(\d{10})$$">
       <action application="bridge" data="freetdm/2/2/$1"/>
      </condition>
 </extension> 

Freetdm.conf file
[span wanpipe wp1]
name => FreeTDM
number => 1
trunk_type => FXS
fxs-channel => 1:5

[span wanpipe wp2]
name => FreeTDM
number => 2
trunk_type => FXO
fxo-channel => 1:1-4

Freetdm.conf.xml file
<configuration name="freetdm.conf" description="FreeTDM Configuration">
  <settings>
    <param name="debug" value="9"/>
    <!--<param name="hold-music" value="$${moh_uri}"/>-->
  </settings>
   <analog_spans>
           <span name="wp1">
                <param name="context" value="context_1"/>
                <param name="dialplan" value="XML"/>
           </span>
            <span name="wp2">
                <param name="context" value="context_1"/>
                <param name="dialplan" value="XML"/>
            </span>
   </analog_spans>
</configuration>

Let me know if I need to provide additional info

Thanks
Sean
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From moises.silva at gmail.com  Wed May 26 14:39:30 2010
From: moises.silva at gmail.com (Moises Silva)
Date: Wed, 26 May 2010 17:39:30 -0400
Subject: [Freeswitch-users] Freetdm error
In-Reply-To: <C822DCD8.B1F5%sean@obscuradigital.com>
References: <C822DCD8.B1F5%sean@obscuradigital.com>
Message-ID: <AANLkTilgQ3iJigTKK3bDkShF2G2Y8gxT5QpgalJOMTKl@mail.gmail.com>

Pastebin the debug output of the FreeSWITCH log.

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com


On Wed, May 26, 2010 at 5:12 PM, Sean Holt <sean at obscuradigital.com> wrote:

>  Hello all,
>
> I?ve been hacking at this issue for the last 3 days and I can?t get beyond
> this error when attempting to call out through my sangoma card.
>
>   SIP/2.0 503 Service Unavailable
>    Via: SIP/2.0/UDP 192.168.100.6:12026
> ;branch=z9hG4bK-d8754z-77564c04563bac12-1---d8754z-;rport=12026
>    From: "Sean Holt"<sip:2028 at 172.16.15.10>;tag=25dacb3d
>    To: <sip:5102079553 at 172.16.15.10>;tag=4St9j3N9yXr8g
>    Call-ID: NWViZTI0ODljMmY1YTM5ZDRiZmQzNTUwYzgwNWVkMDQ.
>    CSeq: 1 INVITE
>    User-Agent: Configured by TCAPI
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
>    Supported: precondition, path, replaces
>    Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
>    Reason: Q.850;cause=34;text="NORMAL_CIRCUIT_CONGESTION"
>    Content-Length: 0
>    Remote-Party-ID: "5102079553" <sip:5102079553 at 172.16.15.10
> >;party=calling;privacy=off;screen=no
>
> Here?s my configuration:
> I?m trying to call out from my internal voip phone through my sangoma card
> b600.  My phone line is plugged into the FXO port.  Module (mod_freetdm) is
> successfully loaded
> Dialplan:
> <extension name="main_number_53">
>       <condition field="destination_number"
> expression="^(\d{11})$|^(\d{10})$$">
>        <action application="bridge" data="freetdm/2/2/$1"/>
>       </condition>
>  </extension>
>
> Freetdm.conf file
> [span wanpipe wp1]
> name => FreeTDM
> number => 1
> trunk_type => FXS
> fxs-channel => 1:5
>
> [span wanpipe wp2]
> name => FreeTDM
> number => 2
> trunk_type => FXO
> fxo-channel => 1:1-4
>
> Freetdm.conf.xml file
> <configuration name="freetdm.conf" description="FreeTDM Configuration">
>   <settings>
>     <param name="debug" value="9"/>
>     <!--<param name="hold-music" value="$${moh_uri}"/>-->
>   </settings>
>    <analog_spans>
>            <span name="wp1">
>                 <param name="context" value="context_1"/>
>                 <param name="dialplan" value="XML"/>
>            </span>
>             <span name="wp2">
>                 <param name="context" value="context_1"/>
>                 <param name="dialplan" value="XML"/>
>             </span>
>    </analog_spans>
> </configuration>
>
> Let me know if I need to provide additional info
>
> Thanks
> Sean
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Wed May 26 14:41:41 2010
From: mike at jerris.com (Michael Jerris)
Date: Wed, 26 May 2010 17:41:41 -0400
Subject: [Freeswitch-users] Direct inward dialling
In-Reply-To: <35FA7A4D-4937-4EA1-8FA4-B124129C92DD@freeswitch.org>
References: <491B305B.4060307@gmx.net>
	<072A175C-9004-4C14-90EC-9A93A8453787@gmail.com>
	<AANLkTinxYIG6S30XMB5GUVuWsXlyhG_slqc_e0PQy0wM@mail.gmail.com>
	<AANLkTinjUuCyrK6eVSalV6fUVNFM-j_zlvrzRxq3lJZQ@mail.gmail.com>
	<AANLkTik0JvNDXax6Ay99hrcgzuiItY0f4LkCsaRjYS8a@mail.gmail.com>
	<AANLkTinYufK3FWtbM95088fOOOWWzOyMpU6ho0VXiOEy@mail.gmail.com>
	<AANLkTinLTWiKrvgPMiiEZArMxMQFCgtJNXoHO_BBlHRN@mail.gmail.com>
	<AANLkTilq_gLLPa-VaRlaiKofqNVlpvm1zgSiL53UTf7z@mail.gmail.com>
	<AANLkTikiraunYdhjcCUmKLmgNAyQ-Sp6r4EpohHuEbZn@mail.gmail.com>
	<AANLkTimMkeVcO7mqzCVA95PE_XSGoGdHfC2A7HJ5i0_R@mail.gmail.com>
	<AANLkTim06fDEoeHe_hVVtfMDUeS2saTi-JyKiJQFB9ZC@mail.gmail.com>
	<35FA7A4D-4937-4EA1-8FA4-B124129C92DD@freeswitch.org>
Message-ID: <BE0E42CB-9888-438F-BBD7-F2CC76FAF7E0@jerris.com>

are these calls to a registering gateway?

On May 26, 2010, at 1:40 AM, Michael S Collins wrote:

> It's all good. Now you have to pay it forward. :)
> -MC
> 
> Sent from my iPhone
> 
> On May 25, 2010, at 9:34 PM, RR <ranjtech at gmail.com> wrote:
> 
>> Michael,
>> 
>> Thank you SO SO much for the help. Your regex work perfectly as desired. I had tried what you suggested earlier but I think I might've made a mistake somewhere because I wasn't getting the right results so I resorted to doing the "|" between the prefixes to strip them out thinking maybe FS works by going if it begins + OR +1 OR 011 then remove them but I guess it doesn't as when "\" appears it uses / matches against only the first one of the those as opposed to all of those. 
>> 
>> Thanks again and sorry for wasting your time ;)
>> 
>> Cheers
>> RR
>> 
>> On Tue, May 25, 2010 at 9:19 PM, Michael Collins <msc at freeswitch.org> wrote:
>> 
>> 
>> On Tue, May 25, 2010 at 5:44 PM, RR <ranjtech at gmail.com> wrote:
>> Michael, haha, yeah they indeed are. That's why I'm routing based on $2, but I still see the 1 and/or the 011 going through to the "bridge" application. Why??
>> Because your regex is wrong. :) It took me a while to figure it out. I'm surprised it worked at all. All the stuff you have inside the first set of parens is not behaving the way you think it should be. If I read your intentions correctly you're trying to strip off leading:
>> +
>> OR
>> +1
>> OR
>> 1
>> 
>> In the first regex. Correct? If ANI is NANPA-ish then try this in your first regex:
>> ^\+?1?([2-9]\d+).*$
>> 
>> That should strip off leading + and/or 1 and capture just the 10-digit phone number in $1. (Be sure to use $1 and not $2, unless you had your heart set on using $2 in which case wrap the first part of the regex in parens)
>> 
>> The other regex is also tricky. I assume you are trying to strip off the same as above as well as 011? Try this:
>> ^\+?1?(011)?([2-9]\d+).*$
>> 
>> Again, if the phone number in question is NANPA then $2 should contain just the 10 digits you want. Play around with that and let us know what happens. Also, don't forget what I said about using regex from the fs_cli. You can test all this stuff yourself. :)
>> 
>> -MC
>> 
>> 
>>  
>> 
>> On Tue, May 25, 2010 at 8:34 PM, Michael Collins <msc at freeswitch.org> wrote:
>> 
>> 
>> On Tue, May 25, 2010 at 5:27 PM, RR <ranjtech at gmail.com> wrote:
>> Ok, so I take that back. This seems to only work when the dialplan has a specific ANI and DNIS / destination_number / sip_to_user defined. If this is more general
>> 
>> like 
>> 
>> <include>
>>   <extension name="public_did">
>>     <condition field="ani" expression="^(\+?|\+1?|1?)(\d+).*$" break="never">
>>         <action application="set" data="effective_caller_id_number=$2"/>
>>         <action application="set" data="effective_caller_id_name=$2"/>
>>     </condition>
>>     <condition field="${sip_to_user}" expression="^(\+1?|\+|1?|011?)(\d+).*$" break="never">
>>       <action application="set" data="continue_on_fail=false"/>
>>       <action application="set" data="hangup_after_bridge=true"/>
>>       <action application="set" data="domain_name=$${domain}"/>
>>       <action application="set" data="bypass_media=true"/>
>>       <action application="limit_hash" data="in cc_blades 4200 !USER_BUSY"/>
>>       <action application="bridge" data="{sip_invite_domain=${sip_from_host}}sofia/gateway/${distributor(cc_blades)}/$2"/>
>>     </condition>
>>   </extension>
>> </include>
>> 
>> then even though the expression/conditions seem to match, none of the digits are being stripped off. Shouldn't this be stripping off digits??
>> 
>> Here's the debug output:
>> 
>> Dialplan: sofia/external/16469NNNNNN Regex (PASS) [public_did] ani(16469NNNNNN;phone-context=+1) =~ /^(\+?|\+1?|1?)(\d+).*$/ break=never
>> Dialplan: sofia/external/16469NNNNNN Action set(effective_caller_id_number=16469NNNNNN)
>> Dialplan: sofia/external/16469NNNNNN Action set(effective_caller_id_name=16469NNNNNN)
>> Dialplan: sofia/external/16469NNNNNN Regex (PASS) [public_did] ${sip_to_user}(011390NNNNNNNNNN;phone-context=+39) =~ /^(\+1?|\+|1?|011?)(\d+).*$/ break=never
>> Dialplan: sofia/external/16469NNNNNN Action set(continue_on_fail=false)
>> Dialplan: sofia/external/16469NNNNNN Action set(hangup_after_bridge=true)
>> Dialplan: sofia/external/16469NNNNNN Action set(domain_name=208.72.186.166)
>> Dialplan: sofia/external/16469NNNNNN Action set(bypass_media=true)
>> Dialplan: sofia/external/16469NNNNNN Action limit_hash(in cc_blades 4200 !USER_BUSY)
>> Dialplan: sofia/external/16469NNNNNN Action bridge({sip_invite_domain=${sip_from_host}}sofia/gateway/${distributor(cc_blades)}/011390NNNNNNNNNN)
>> 
>> why're the '1' in the ANI and '011' in the DNIS/sip_to_user being stripped off???
>> 
>> Regex 101 :)
>> 
>> The 1 or the 011 are in $1
>> -MC
>> 

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From bobc at devassert.com  Wed May 26 14:56:08 2010
From: bobc at devassert.com (Bob Coleman)
Date: Thu, 27 May 2010 09:56:08 +1200
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
In-Reply-To: <4BFD4562.5030804@gmail.com>
References: <490aedfdd7c8ac680bbe0d7f68c786de@mail.gmail.com>
	<4BFBE6C2.5000703@gmail.com>
	<AANLkTilsHn9znajzF0Jj-QDu4je9QW7CeymuBYE0SyKa@mail.gmail.com>
	<4BFBEF7C.9020107@gmail.com>
	<9250A9FF05DB4B879DEF6ECE8E3FFAC1@dell9400>
	<AANLkTilU2OwgH0Qk_Qoc3R7ErPO_Z_VRhePusetkRkeW@mail.gmail.com>
	<AANLkTikXWvEB_C3jqOWLHaVcns9-U5Hok-5MqgE_uzzZ@mail.gmail.com>
	<4BFC4A64.5090208@gmail.com>
	<AANLkTinFtVXa7CPou-ShPYIXHq0AktokhptGSEFKBkge@mail.gmail.com>
	<D311BDCB06324D0DB77898FC6FF4B0EB@dell9400>
	<AANLkTilAMe1lRKK4R6Nmr29FzfI-pP3wRa7Foabmd8Dp@mail.gmail.com>
	<4BFD4562.5030804@gmail.com>
Message-ID: <AANLkTikGQlZ4N3zfYv18COvbubEi-8LSOZxnAy1KC5sP@mail.gmail.com>

Hi,

Will incorporate some threading into the example I am working on for you.

The outbound event socket method is very similar to the dialogic
environment, I know how you felt though, but by starting small(like
just even answering a call) gets you moving pretty quick. Freeswitch
is lots of fun to work with, and the guys on here are very
supportive!!

Bob

On Thu, May 27, 2010 at 3:59 AM, Phone <phone.bytes at gmail.com> wrote:
> Thanks to all for the most helpful feedback. ?Sharing your approaches
> and experiences are a big help. ?I look forward to the upcoming code
> samples.
>
> I was coming from a windows/dialogic environment where I used a library
> that allowed me to work on a little higher level. ?For example, I had a
> call to "play a file" that took a parameter of whether or not to allow a
> dtmf to interrupt. ?There was also a call to "ReadDtmfs" that took
> parameters to specify the number of Dtmf's to read, how long to wait for
> them, and what terminating character to use. ?I guess that you could
> write some scripts or compiled code with these same types of functions
> to simplify some of these routine tasks with reusable code?
>
> Also, the library handled the threading and scheduling with the OS. ?I
> am still unclear on handling the events. ?I guess you have a big loop
> reading events and then acting on them using the uuid to determine which
> call it is and how to deal with the next step of the call? ?Any feedback
> on this part of the project?
>
> Again, Thanks!
>
> Bob Coleman wrote:
>> Ah sorry, I started with the esl to get an understanding then wrote my
>> own socket library(was actually very easy to do), when I mean docs I
>> mean the event socket docs. I still think of it as the esl, my
>> mistake.
>>
>> http://wiki.freeswitch.org/wiki/Event_Socket
>>
>> I started with a codeplex project, that had been abandoned, and then
>> once I understood the structure of the event socket language, was able
>> to rewrite it to better handle what we were doing.
>>
>> I also married it up to an old gotdotnet asterisk fast agi project,
>> once again abandoned, to allow for the use of asterisk as well, but in
>> the end freeswitch won because we could use just one platform.
>>
>> I am busy writing a small sample app at the moment to demonstrate a
>> problem I am trying to solve. Can release that code once sorted. Will
>> be in a week or so. Am intending it as a quick way of testing event
>> sockets, and trying various commands etc. before commiting to coding
>> something.
>>
>> Bob
>>
>> On Wed, May 26, 2010 at 2:26 PM, Jan Berger <jan.berger at video24.no> wrote:
>>
>>> Hi,
>>>
>>> Do you have some sample code you could share + what docs did you look at?
>>>
>>> I would like to write and test some C# using ESL for my own work.
>>>
>>> Jan
>>>
>>> -----Original Message-----
>>> From: freeswitch-users-bounces at lists.freeswitch.org
>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bob
>>> Coleman
>>> Sent: 26. mai 2010 04:07
>>> To: freeswitch-users at lists.freeswitch.org
>>> Subject: Re: [Freeswitch-users] Questions on Building an application for
>>> FreeSWITCH
>>>
>>> We used c# as the rest of our systems are windows based. The language
>>> doesnt matter too much, as long as you know where you are headed, what
>>> performance you require, and what platform you are going to be using.
>>>
>>> Found the ESL so much easier than the dialogic c library we were using.
>>>
>>> The docs for the esl are easy to understand, the thing I couldnt get
>>> my head around initially was the dialing out, with the dialogic you
>>> are in the middle when you dial, ie already on the channel, but with
>>> freeswitch you are kind of the third party when you dial, the channel
>>> being created by the dialing and handing it off to be worked on. We
>>> make the call via an inbound event socket and hand it off to an
>>> outbound event socket application via the dialplan.
>>>
>>> On Wed, May 26, 2010 at 10:08 AM, Phone <phone.bytes at gmail.com> wrote:
>>>
>>>> Thanks for the info. ?What language did you use?
>>>>
>>>> Bob Coleman wrote:
>>>>
>>>>> Have just recently completed a project to convert an old windows
>>>>> dialogic application(vb6) to FreeSWITCH, would reccommend using the
>>>>> ESL, was able to map the old dialogic calls to the ESL calls pretty
>>>>> easily. We used a mixture of inbound and outbound sockets, as we have
>>>>> people dialing us, not just dialing out etc.
>>>>>
>>>>> With the dialogic you open a port and make the call and handle the
>>>>> dtmf, with freeswitch you create a socket connection to FreeSWITCH to
>>>>> dial the number and then hand it off to an extension for processing
>>>>> the dtmf(that is one approach any way)
>>>>>
>>>>> Bob
>>>>>
>>>>> On Wed, May 26, 2010 at 6:45 AM, Michael Collins <msc at freeswitch.org>
>>>>>
>>> wrote:
>>>
>>>>>> On Tue, May 25, 2010 at 11:38 AM, Jan Berger <jan.berger at video24.no>
>>>>>>
>>> wrote:
>>>
>>>>>>> Actually - before you get "to smart" - may I suggest that you start
>>>>>>> writing
>>>>>>> - or improving - the getting started sections of the doc. Address the
>>>>>>> areas
>>>>>>> where you struggle and let others benefit from your work.
>>>>>>>
>>>>>>> I have been through similar issues myself - FS is one of the easier
>>>>>>> projects
>>>>>>> to work with once you get under the hood, but you basically need to
>>>>>>>
>>> evolve
>>>
>>>>>>> to the level where you read the source code.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>> And if you can wait 2+ months for "the book" then that should help as
>>>>>>
>>> well.
>>>
>>>>>> :D
>>>>>> -MC
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From gerrit308 at gmail.com  Wed May 26 14:49:31 2010
From: gerrit308 at gmail.com (humbr)
Date: Wed, 26 May 2010 14:49:31 -0700 (PDT)
Subject: [Freeswitch-users] Searching mailing lists
In-Reply-To: <0A8EE721-1247-4E78-9B99-BEFB156799A6@freeswitch.org>
References: <1229079085.100.1274717262156.JavaMail.root@srvr12.remotelinkml.com>
	<0A8EE721-1247-4E78-9B99-BEFB156799A6@freeswitch.org>
Message-ID: <1274910571528-5105924.post@n2.nabble.com>


You can also use: http://freeswitch-users.2379917.n2.nabble.com/ 
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Searching-mailing-lists-tp5095136p5105924.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From sean at obscuradigital.com  Wed May 26 15:17:42 2010
From: sean at obscuradigital.com (Sean Holt)
Date: Wed, 26 May 2010 15:17:42 -0700
Subject: [Freeswitch-users] Freetdm error
In-Reply-To: <AANLkTilgQ3iJigTKK3bDkShF2G2Y8gxT5QpgalJOMTKl@mail.gmail.com>
Message-ID: <C822EC16.B205%sean@obscuradigital.com>

Here you go

http://pastebin.freeswitch.org/13057

Thanks


On 5/26/10 2:39 PM, "Moises Silva" <moises.silva at gmail.com> wrote:

> Pastebin the debug output of the FreeSWITCH log.
> 
> Moises Silva
> Senior Software Engineer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
> Canada
> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
> 
> 
> On Wed, May 26, 2010 at 5:12 PM, Sean Holt <sean at obscuradigital.com> wrote:
>> Hello all,
>> 
>> I?ve been hacking at this issue for the last 3 days and I can?t get beyond
>> this error when attempting to call out through my sangoma card.
>> 
>> ??SIP/2.0 503 Service Unavailable
>> ???Via: SIP/2.0/UDP
>> 192.168.100.6:12026;branch=z9hG4bK-d8754z-77564c04563bac12-1---d8754z-;rport=
>> 12026
>> ???From: "Sean Holt"<sip:2028 at 172.16.15.10 <http://2028 at 172.16.15.10>
>> >;tag=25dacb3d
>> ???To: <sip:5102079553 at 172.16.15.10 <http://5102079553 at 172.16.15.10>
>> >;tag=4St9j3N9yXr8g
>> ???Call-ID: NWViZTI0ODljMmY1YTM5ZDRiZmQzNTUwYzgwNWVkMDQ.
>> ???CSeq: 1 INVITE
>> ???User-Agent: Configured by TCAPI
>> ???Accept: application/sdp
>> ???Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
>> REFER, NOTIFY, PUBLISH, SUBSCRIBE
>> ???Supported: precondition, path, replaces
>> ???Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
>> include-session-description, presence.winfo, message-summary, refer
>> ???Reason: Q.850;cause=34;text="NORMAL_CIRCUIT_CONGESTION"
>> ???Content-Length: 0
>> ???Remote-Party-ID: "5102079553" <sip:5102079553 at 172.16.15.10
>> <http://5102079553 at 172.16.15.10> >;party=calling;privacy=off;screen=no
>> 
>> Here?s my configuration:
>> I?m trying to call out from my internal voip phone through my sangoma card
>> b600. ?My phone line is plugged into the FXO port. ?Module (mod_freetdm) is
>> successfully loaded
>> Dialplan:
>> <extension name="main_number_53">
>> ??????<condition field="destination_number"
>> expression="^(\d{11})$|^(\d{10})$$">
>> ???????<action application="bridge" data="freetdm/2/2/$1"/>
>> ??????</condition>
>> ?</extension> 
>> 
>> Freetdm.conf file
>> [span wanpipe wp1]
>> name => FreeTDM
>> number => 1
>> trunk_type => FXS
>> fxs-channel => 1:5
>> 
>> [span wanpipe wp2]
>> name => FreeTDM
>> number => 2
>> trunk_type => FXO
>> fxo-channel => 1:1-4
>> 
>> Freetdm.conf.xml file
>> <configuration name="freetdm.conf" description="FreeTDM Configuration">
>> ??<settings>
>> ????<param name="debug" value="9"/>
>> ????<!--<param name="hold-music" value="$${moh_uri}"/>-->
>> ??</settings>
>> ???<analog_spans>
>> ???????????<span name="wp1">
>> ????????????????<param name="context" value="context_1"/>
>> ????????????????<param name="dialplan" value="XML"/>
>> ???????????</span>
>> ????????????<span name="wp2">
>> ????????????????<param name="context" value="context_1"/>
>> ????????????????<param name="dialplan" value="XML"/>
>> ????????????</span>
>> ???</analog_spans>
>> </configuration>
>> 
>> Let me know if I need to provide additional info
>> 
>> Thanks
>> Sean 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From codecomplete at free.fr  Wed May 26 15:49:58 2010
From: codecomplete at free.fr (GillesToo)
Date: Wed, 26 May 2010 15:49:58 -0700 (PDT)
Subject: [Freeswitch-users] [Out of curiosity] Why not use a voice modem
	for FXO?
In-Reply-To: <AANLkTimvJNT2NPp63C-IeiII-BRTti_CNoWJJjKUmZVS@mail.gmail.com>
References: <1274881028458-5103502.post@n2.nabble.com>
	<6596AAE7-B489-4D4A-B661-EDE2CD351DC3@freeswitch.org>
	<1274882347926-5103647.post@n2.nabble.com>
	<AANLkTikeqyqds4FVor8qBZ9PZgrz9dIPATPAbr2N8TCQ@mail.gmail.com>
	<1274905609840-5105604.post@n2.nabble.com>
	<AANLkTimvJNT2NPp63C-IeiII-BRTti_CNoWJJjKUmZVS@mail.gmail.com>
Message-ID: <1274914198334-5106158.post@n2.nabble.com>



Anthony Minessale wrote:
> A FXO card works as well as it can but traditionally requires an
> echo canceler if you want quality that will not draw complaints

Why do FXO cards generate echo issues while ATA don't? For instance, there
are a lot of people complaining about echo with the Linksys 3102.


Anthony Minessale wrote:
> The good thing about an ATA is that it's self-contained and translates
> everything into SIP for you when you don't have room for hardware.

Actually, that's why I prefer to use a PCI card instead of an ATA: No need
for a transformer and a cable to connect the ATA to the Freeswitch server,
and only one cable left to connect the PCI card to the wall plug. With
non-techie customers, it seems like a safer solution.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Out-of-curiosity-Why-not-use-a-voice-modem-for-FXO-tp5103502p5106158.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Wed May 26 15:56:25 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 26 May 2010 17:56:25 -0500
Subject: [Freeswitch-users] [Out of curiosity] Why not use a voice modem
	for FXO?
In-Reply-To: <1274914198334-5106158.post@n2.nabble.com>
References: <1274881028458-5103502.post@n2.nabble.com>
	<6596AAE7-B489-4D4A-B661-EDE2CD351DC3@freeswitch.org>
	<1274882347926-5103647.post@n2.nabble.com>
	<AANLkTikeqyqds4FVor8qBZ9PZgrz9dIPATPAbr2N8TCQ@mail.gmail.com>
	<1274905609840-5105604.post@n2.nabble.com>
	<AANLkTimvJNT2NPp63C-IeiII-BRTti_CNoWJJjKUmZVS@mail.gmail.com>
	<1274914198334-5106158.post@n2.nabble.com>
Message-ID: <AANLkTilvu9N-BIf3FlQhTGjjFqInrsOZYrGT77bcvO99@mail.gmail.com>

ok,

sure, we support a variety of cards.  I personally recommend sangoma.


On Wed, May 26, 2010 at 5:49 PM, GillesToo <codecomplete at free.fr> wrote:

>
>
> Anthony Minessale wrote:
> > A FXO card works as well as it can but traditionally requires an
> > echo canceler if you want quality that will not draw complaints
>
> Why do FXO cards generate echo issues while ATA don't? For instance, there
> are a lot of people complaining about echo with the Linksys 3102.
>
>
> Anthony Minessale wrote:
> > The good thing about an ATA is that it's self-contained and translates
> > everything into SIP for you when you don't have room for hardware.
>
> Actually, that's why I prefer to use a PCI card instead of an ATA: No need
> for a transformer and a cable to connect the ATA to the Freeswitch server,
> and only one cable left to connect the PCI card to the wall plug. With
> non-techie customers, it seems like a safer solution.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Out-of-curiosity-Why-not-use-a-voice-modem-for-FXO-tp5103502p5106158.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
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
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From codecomplete at free.fr  Wed May 26 16:09:59 2010
From: codecomplete at free.fr (GillesToo)
Date: Wed, 26 May 2010 16:09:59 -0700 (PDT)
Subject: [Freeswitch-users] [Out of curiosity] Why not use a voice modem
	for FXO?
In-Reply-To: <AANLkTilvu9N-BIf3FlQhTGjjFqInrsOZYrGT77bcvO99@mail.gmail.com>
References: <1274881028458-5103502.post@n2.nabble.com>
	<6596AAE7-B489-4D4A-B661-EDE2CD351DC3@freeswitch.org>
	<1274882347926-5103647.post@n2.nabble.com>
	<AANLkTikeqyqds4FVor8qBZ9PZgrz9dIPATPAbr2N8TCQ@mail.gmail.com>
	<1274905609840-5105604.post@n2.nabble.com>
	<AANLkTimvJNT2NPp63C-IeiII-BRTti_CNoWJJjKUmZVS@mail.gmail.com>
	<1274914198334-5106158.post@n2.nabble.com>
	<AANLkTilvu9N-BIf3FlQhTGjjFqInrsOZYrGT77bcvO99@mail.gmail.com>
Message-ID: <1274915399680-5106229.post@n2.nabble.com>


Thanks for the feeback.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Out-of-curiosity-Why-not-use-a-voice-modem-for-FXO-tp5103502p5106229.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From codeghar at gmail.com  Wed May 26 18:40:59 2010
From: codeghar at gmail.com (Code Ghar)
Date: Wed, 26 May 2010 20:40:59 -0500
Subject: [Freeswitch-users] CDRs for Gateways
Message-ID: <AANLkTimj6QaSqIGzsZ-hLn9VIjhpBCPiWrsHYHWMZv21@mail.gmail.com>

First off, I want to apologize for breaking up this thread into three
threads. This is the first time I have ever used mailing lists and it's
taken me some time to get things right. Now things should work as they must.

David, in response to your last message, in my hurry I forgot to make the
necessary modifications in the copy-pasting of config. Here it is rectified.

---Before---
<!-- Only log in Master.csv -->
<!--<param name="master-file-only" value="true"/>-->

---After---
<!-- Only log in Master.csv -->
<param name="master-file-only" value="true"/>

So all I did was enable the master file. After reloadxml all CDRs are added
to master file, including those of gateways.






Well, according to your mail, you did not change anything :)
I guess it is a typo.

Can you confirm what you changed ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
? l'intention exclusive de ses destinataires. Toute utilisation ou
diffusion non autoris?e est interdite. Tout message ?lectronique est
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes
pas destinataire de ce message, merci de le d?truire imm?diatement et
d'avertir l'exp?diteur.




Le 22/05/2010 ? 03:17, Code Ghar a ?crit :

> Hi David
>
> I looked at what you suggested. Before your suggestion, in /usr/
> local/freeswitch/conf/autoload_configs/cdr_csv.conf file, I had the
> following lines
>
>
> <!-- Only log in Master.csv -->
> <!--<param name="master-file-only" value="true"/>-->
>
> After your suggestion, I changed it to
>
> <!-- Only log in Master.csv -->
> <!--<param name="master-file-only" value="true"/>-->
>
> and then did a reloadxml. Now all CDRs were created in /usr/local/
> freeswitch/log/cdr-csv/Master.csv and CDRs for inbound calls from
> unregistered sources (i.e. external gateways) were also created in
> this file. Thanks for your help.
>
>
>
> Code,
>
> I think you have a major issue or configuration error with your FS
> setup.
> FS, as any B2BUA, generates a CDR for any inbound leg, either from a
> registered user or from a non-registered gateway.
> It doesn't care where it comes from, it just does it.
> I can barely imagine a softswitch that would not write the CDR
> matching a call that was routed through it.
>
> The CDR for the outbound leg is optional.
>
> Can you confirm you are looking in Master.csv ?
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
> ? l'intention exclusive de ses destinataires. Toute utilisation ou
> diffusion non autoris?e est interdite. Tout message ?lectronique est
> susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre
> de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes
> pas destinataire de ce message, merci de le d?truire imm?diatement et
> d'avertir l'exp?diteur.
>
>
>
>
> Le 21/05/2010 ? 05:25, Code Ghar a ?crit :
>
> > Using mod_cdr_csv, it is easy to create CDRs for registered users.
> > However, how can we get CDRs for external gateways? For example,
> > FreeSWITCH is configured to receive calls from one gateway and send
> > them out through another. In this scenario, when neither inbound nor
> > outbound gateway registers itself to FreeSWITCH, how can we generate
> > and store CDRs for inbound gateway?
> >
> > Similarly, let's say a SIP provider sends a call meant for a
> > registered user. For example, if a cell phone calls 6175550000, it
> > is routed through the telephone network to the SIP provider. This
> > provider routes the call to FreeSWITCH, which looks up in its dial
> > plan that this number should be routed to extension (or registered
> > user) 1000. There are no CDRs generated for either the SIP
> > provider's gateway or the registered user in this case.
> >
> > In both these scenarios, if a call is inbound to FreeSWITCH but has
> > not been generated by a registered user, mod_cdr_csv does not create
> > CDRs. I have even tried to set the following parameter in
> > cdr_csv.conf.xml      but it didn't help.
> >
> > <param name="legs" value="ab"/>
> >
> > My question is this: can mod_cdr_csv generate CDRs for inbound calls
> > from sources other than registered users? If it can't, how can this
> > be achieved?
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
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>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From william.suffill at gmail.com  Wed May 26 18:58:07 2010
From: william.suffill at gmail.com (William Suffill)
Date: Wed, 26 May 2010 21:58:07 -0400
Subject: [Freeswitch-users] Searching mailing lists
In-Reply-To: <1274910571528-5105924.post@n2.nabble.com>
References: <1229079085.100.1274717262156.JavaMail.root@srvr12.remotelinkml.com>
	<0A8EE721-1247-4E78-9B99-BEFB156799A6@freeswitch.org>
	<1274910571528-5105924.post@n2.nabble.com>
Message-ID: <AANLkTimTQPnFWDGTvxpfaJI8rhfpNfQR5ZINF2Jn3_gn@mail.gmail.com>

http://dir.gmane.org/search.php?match=freeswitch works too for the FS
lists.
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From codeghar at gmail.com  Wed May 26 19:01:39 2010
From: codeghar at gmail.com (Code Ghar)
Date: Wed, 26 May 2010 21:01:39 -0500
Subject: [Freeswitch-users] FS as Media Gateway Only
Message-ID: <AANLkTimgKRtL06xmyjXUPnpNIjlYkHC6BytzmTqKmVm7@mail.gmail.com>

Is it possible -- and are there any case studies, practical experience, etc
-- on deploying FreeSWITCH (FS) in this architecture: one server (FSSIP)
handles SIP signaling only, and multiple servers (FSRTP1, FSRTP2, ...,
FSRTPn) handle all media responsibilities? So when a call comes in, the SDP
contains IP of, say FSRTP1, as media handler. For this to work, FSSIP would
request FSRTPx for media resources for each new call and add its IP and port
in SDP. The media servers/gateways would play IVR, etc.; collect DTMF and
forward as appropriate to FSSIP; perform transcoding; etc.; all while FSSIP
only deals with signaling. This way multiple servers could be deployed to
handle media responsibilities and only a handful would be required for
signaling. In future if there's a greater need for transcoding, etc. all you
need to do is deploy a media server and not have to add servers for
signaling.

This idea came to me because I have come across two proprietary applications
that do it this way. They have a SIP component and a media component. You
can run both on the same physical machine or you can separate them out into
multiple machines.

Another way for this could be to integrate FS as a media component to
another application's SIP component. A mix-and-match, so to speak.

On the flip side, deploy FS as a SIP server and use media capabilities of
some other hardware or software application. For example, FS handles
signaling and use dedicated hardware for media. A good example of this is
illustrated (somewhat) by an image on Sangoma's website:
http://www.sangoma.com/assets/images/content/transcoding_diagram.jpg. Look
at the "pooled transcoding".

Is FS even designed to be this modular? If so, how can the aforementioned
scenario(s) be achieved?
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From brian at freeswitch.org  Wed May 26 19:26:05 2010
From: brian at freeswitch.org (Brian West)
Date: Wed, 26 May 2010 21:26:05 -0500
Subject: [Freeswitch-users] Searching mailing lists
In-Reply-To: <AANLkTimTQPnFWDGTvxpfaJI8rhfpNfQR5ZINF2Jn3_gn@mail.gmail.com>
References: <1229079085.100.1274717262156.JavaMail.root@srvr12.remotelinkml.com>
	<0A8EE721-1247-4E78-9B99-BEFB156799A6@freeswitch.org>
	<1274910571528-5105924.post@n2.nabble.com>
	<AANLkTimTQPnFWDGTvxpfaJI8rhfpNfQR5ZINF2Jn3_gn@mail.gmail.com>
Message-ID: <C854AED1-27A2-4540-82A0-F7ED71B5DED5@freeswitch.org>

what is so wrong with using your google foo?  include site:lists.freeswitch.org in your search box and it'll limit the search to lists.freeswitch.org which is easy to do :P

/b

On May 26, 2010, at 8:58 PM, William Suffill wrote:

> http://dir.gmane.org/search.php?match=freeswitch works too for the FS lists.  _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From sid at eltc.ru  Wed May 26 19:27:56 2010
From: sid at eltc.ru (Sergey Scheglov)
Date: Thu, 27 May 2010 09:27:56 +0700
Subject: [Freeswitch-users] Don't work playback after bypass media mode.
In-Reply-To: <23EA73D6-FAF2-4067-84E0-6A2DDCB339E7@freeswitch.org>
References: <1427456515.20100526174916@eltc.ru>
	<23EA73D6-FAF2-4067-84E0-6A2DDCB339E7@freeswitch.org>
Message-ID: <82209492.20100527092756@eltc.ru>

Hello, Brian.

You wrote 27 may 2010, 2:52:05:

> What rev are you running?

> /b

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

Thank's for reply, Brian

[root at freeswitch]# fs_cli -x "version"
FreeSWITCH Version 1.0.head (git-)

:(
I don't known why/
But "git clone" from 05/20/2010.

-- 
Regard
Sergey Scheglov




From william.suffill at gmail.com  Wed May 26 19:34:51 2010
From: william.suffill at gmail.com (William Suffill)
Date: Wed, 26 May 2010 22:34:51 -0400
Subject: [Freeswitch-users] Searching mailing lists
In-Reply-To: <C854AED1-27A2-4540-82A0-F7ED71B5DED5@freeswitch.org>
References: <1229079085.100.1274717262156.JavaMail.root@srvr12.remotelinkml.com>
	<0A8EE721-1247-4E78-9B99-BEFB156799A6@freeswitch.org>
	<1274910571528-5105924.post@n2.nabble.com> 
	<AANLkTimTQPnFWDGTvxpfaJI8rhfpNfQR5ZINF2Jn3_gn@mail.gmail.com> 
	<C854AED1-27A2-4540-82A0-F7ED71B5DED5@freeswitch.org>
Message-ID: <AANLkTinz9-2khXKZ4DYJ7z7Gu_M2lomJ6Rac6N9l71UF@mail.gmail.com>

Nothing just offering options. Everyone has a personal preference. I happen
to just subscribe to the lists on my GMAIL and call it a day. =)

-- W
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From sid at eltc.ru  Wed May 26 21:48:20 2010
From: sid at eltc.ru (Sergey Scheglov)
Date: Thu, 27 May 2010 11:48:20 +0700
Subject: [Freeswitch-users] Don't work playback after bypass media mode.
In-Reply-To: <23EA73D6-FAF2-4067-84E0-6A2DDCB339E7@freeswitch.org>
References: <1427456515.20100526174916@eltc.ru>
	<23EA73D6-FAF2-4067-84E0-6A2DDCB339E7@freeswitch.org>
Message-ID: <1235356585.20100527114820@eltc.ru>

Hi, Brian.

You wrote  27 may 2010 ?., 2:52:05:

> What rev are you running?

> /b

> On May 26, 2010, at 5:49 AM, Sergey Scheglov wrote:

Thank's for reply.

I'm update git and FS.

Now,
[root at freeswitch]# fs_cli -x "version"
FreeSWITCH Version 1.0.head (git-095815f 2010-05-26 20-58-16 -0500)

The problem still remains.


Regard's
Sergey Scheglov




From mike at jerris.com  Wed May 26 22:25:34 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 27 May 2010 01:25:34 -0400
Subject: [Freeswitch-users] libtool version mismatch
In-Reply-To: <640313454.80.1274894541888.JavaMail.root@srvr12.remotelinkml.com>
References: <640313454.80.1274894541888.JavaMail.root@srvr12.remotelinkml.com>
Message-ID: <5C369850-45CE-4BB8-B2AC-635919D7385B@jerris.com>

if you checked out from git and have no local changes, you can use

git clean -f -d -x

if you downloaded a Tarball, delete the whole source tree and re  
extract the tarball

On May 26, 2010, at 1:22 PM, David Swardstrom <dswardstrom at remotelink.com 
 > wrote:

> I think I understand what "wipe your tree" means in a sense.
> But I don't know what I need to do this.
> I suspect that this means the I need to delete certain files in every
> subdirectory but what files.
>
> While I have used Unix based systems in the past, I never had to use  
> make
> because either some other build process was used or make was embedded
> in other tools. For the last couple of years, I have been working
> with Visual Studio and C++.
>
> I think that this "clean-up" everything related to the make and
> remake would be something that could be useful to add to one of the
> Wiki pages.
>
> Regards,
> Paul David Swardstrom
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From mike at jerris.com  Wed May 26 22:30:25 2010
From: mike at jerris.com (Michael Jerris)
Date: Thu, 27 May 2010 01:30:25 -0400
Subject: [Freeswitch-users] Don't work playback after bypass media mode.
In-Reply-To: <1427456515.20100526174916@eltc.ru>
References: <1427456515.20100526174916@eltc.ru>
Message-ID: <C7E0DD6C-0EBB-4C1F-A47A-D3685DF5C6FC@jerris.com>

we can't tell why because you omitted the part of the log that may  
show why.  also, why do you set continue on fail twice in a row?

On May 26, 2010, at 6:49 AM, Sergey Scheglov <sid at eltc.ru> wrote:

> Hi, Freeswitch-users.
>
> My dialplan:
> <extension name="Local_Extension">
>    <condition field="destination_number" expression="^([1-9]\d{4})$">
>        <action application="set" data="dialed_extension=$1"/>
>        <action application="export" data="dialed_extension=$1"/>
>        <action application="set" data="ringback=${ru-ring}"/>
>        <action application="set" data="bypass_media=true"/>
>        <action application="set" data="hangup_after_bridge=true"/>
>        <action application="set"  
> data="continue_on_fail=SUBSCRIBER_ABSENT,USER_NOT_REGISTERED"/>
>        <action application="set" data="continue_on_fail=true"/>
>        <action application="bridge" data="user/${dialed_extension}@$ 
> {domain_name}"/>
>        <action application="pre_answer"/>
>        <action application="sleep" data="1000"/>
>        <action application="playback" data="elight/$ 
> {originate_disposition}.wav"/>
>    </condition>
> </extension>
>
> If hangup cause is SUBSCRIBER_ABSENT or USER_NOT_REGISTERED, then
> playback don't work.
>
> Piece of log
>
> 2010-05-26 17:35:08.218253 [ERR] switch_ivr_originate.c:2493 Cannot  
> create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT]
> 2010-05-26 17:35:08.218253 [DEBUG] switch_ivr_originate.c:3308  
> Originate Resulted in Error Cause: 20 [SUBSCRIBER_ABSENT]
> 2010-05-26 17:35:08.218253 [INFO] mod_dptools.c:2356 Originate  
> Failed.  Cause: SUBSCRIBER_ABSENT
> EXECUTE sofia/internal/39663 at 10.110.1.8:5060 pre_answer()
> 2010-05-26 17:35:08.219282 [INFO] mod_dptools.c:746 Sending early  
> media
> 2010-05-26 17:35:08.219282 [NOTICE] mod_sofia.c:2032 Pre-Answer  
> sofia/internal/39663 at 10.110.1.8:5060!
> 2010-05-26 17:35:08.219282 [DEBUG] switch_core_session.c:642 Send  
> signal sofia/internal/39663 at 10.110.1.8:5060 [BREAK]
> EXECUTE sofia/internal/39663 at 10.110.1.8:5060 sleep(1000)
> 2010-05-26 17:35:08.219282 [DEBUG] sofia.c:4195 Channel sofia/ 
> internal/39663 at 10.110.1.8:5060 skipping state [early][183]
> 2010-05-26 17:35:09.219252 [DEBUG] switch_core_session.c:642 Send  
> signal sofia/internal/39663 at 10.110.1.8:5060 [BREAK]
> 2010-05-26 17:35:39.226271 [ERR] switch_core_io.c:121 sofia/internal/ 
> 39663 at 10.110.1.8:5060 has no read codec.
> 2010-05-26 17:35:39.226271 [DEBUG] switch_core_session.c:1759  
> Application playback Requires media on channel sofia/internal/ 
> 39663 at 10.110.1.8:5060!
> EXECUTE sofia/internal/39663 at 10.110.1.8:5060 playback(elight/ 
> SUBSCRIBER_ABSENT.wav)
> 2010-05-26 17:35:39.226271 [NOTICE] switch_core_state_machine.c:185  
> sofia/internal/39663 at 10.110.1.8:5060 has executed the last dialplan  
> instruction, hanging up.
> 2010-05-26 17:35:39.226271 [NOTICE] switch_core_state_machine.c:187  
> Hangup sofia/internal/39663 at 10.110.1.8:5060 [CS_EXECUTE]  
> [NORMAL_CLEARING]
>
> What am I doing wrong?
>
> If set bypass_media_after_bridge instead bypass_media, then works
> fine, BUT changing codec negotiation.
>
> Thanks.
> -- 
> Regard
> Sergey Scheglov
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From david.ponzone at gmail.com  Wed May 26 22:43:02 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Thu, 27 May 2010 07:43:02 +0200
Subject: [Freeswitch-users] CDRs for Gateways
In-Reply-To: <AANLkTimj6QaSqIGzsZ-hLn9VIjhpBCPiWrsHYHWMZv21@mail.gmail.com>
References: <AANLkTimj6QaSqIGzsZ-hLn9VIjhpBCPiWrsHYHWMZv21@mail.gmail.com>
Message-ID: <7F02B782-75C0-43E0-8AE7-79ED89157D0F@gmail.com>

Ok, but you should get the CDRs also with master-file-only commented.
It's my configuration on all my servers.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 27/05/2010 ? 03:40, Code Ghar a ?crit :

> First off, I want to apologize for breaking up this thread into  
> three threads. This is the first time I have ever used mailing lists  
> and it's taken me some time to get things right. Now things should  
> work as they must.
>
> David, in response to your last message, in my hurry I forgot to  
> make the necessary modifications in the copy-pasting of config. Here  
> it is rectified.
>
> ---Before---
> <!-- Only log in Master.csv -->
> <!--<param name="master-file-only" value="true"/>-->
>
> ---After---
> <!-- Only log in Master.csv -->
> <param name="master-file-only" value="true"/>
>
> So all I did was enable the master file. After reloadxml all CDRs  
> are added to master file, including those of gateways.
>
>
>
>
>
>
> Well, according to your mail, you did not change anything :)
> I guess it is a typo.
>
> Can you confirm what you changed ?
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
> ? l'intention exclusive de ses destinataires. Toute utilisation ou
> diffusion non autoris?e est interdite. Tout message ?lectronique est
> susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre
> de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes
> pas destinataire de ce message, merci de le d?truire imm?diatement et
> d'avertir l'exp?diteur.
>
>
>
>
> Le 22/05/2010 ? 03:17, Code Ghar a ?crit :
>
> > Hi David
> >
> > I looked at what you suggested. Before your suggestion, in /usr/
> > local/freeswitch/conf/autoload_configs/cdr_csv.conf file, I had the
> > following lines
> >
> >
> > <!-- Only log in Master.csv -->
> > <!--<param name="master-file-only" value="true"/>-->
> >
> > After your suggestion, I changed it to
> >
> > <!-- Only log in Master.csv -->
> > <!--<param name="master-file-only" value="true"/>-->
> >
> > and then did a reloadxml. Now all CDRs were created in /usr/local/
> > freeswitch/log/cdr-csv/Master.csv and CDRs for inbound calls from
> > unregistered sources (i.e. external gateways) were also created in
> > this file. Thanks for your help.
> >
> >
> >
> > Code,
> >
> > I think you have a major issue or configuration error with your FS
> > setup.
> > FS, as any B2BUA, generates a CDR for any inbound leg, either from a
> > registered user or from a non-registered gateway.
> > It doesn't care where it comes from, it just does it.
> > I can barely imagine a softswitch that would not write the CDR
> > matching a call that was routed through it.
> >
> > The CDR for the outbound leg is optional.
> >
> > Can you confirm you are looking in Master.csv ?
> >
> > David Ponzone  Direction Technique
> > email: david.ponzone at ipeva.fr
> > tel:      01 74 03 18 97
> > gsm:   06 66 98 76 34
> >
> > Service Client IPeva
> > tel:      0811 46 26 26
> > www.ipeva.fr  -   www.ipeva-studio.com
> >
> > Ce message et toutes les pi?ces jointes sont confidentiels et  
> ?tablis
> > ? l'intention exclusive de ses destinataires. Toute utilisation ou
> > diffusion non autoris?e est interdite. Tout message ?lectronique est
> > susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
> titre
> > de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes
> > pas destinataire de ce message, merci de le d?truire imm?diatement  
> et
> > d'avertir l'exp?diteur.
> >
> >
> >
> >
> > Le 21/05/2010 ? 05:25, Code Ghar a ?crit :
> >
> > > Using mod_cdr_csv, it is easy to create CDRs for registered users.
> > > However, how can we get CDRs for external gateways? For example,
> > > FreeSWITCH is configured to receive calls from one gateway and  
> send
> > > them out through another. In this scenario, when neither inbound  
> nor
> > > outbound gateway registers itself to FreeSWITCH, how can we  
> generate
> > > and store CDRs for inbound gateway?
> > >
> > > Similarly, let's say a SIP provider sends a call meant for a
> > > registered user. For example, if a cell phone calls 6175550000, it
> > > is routed through the telephone network to the SIP provider. This
> > > provider routes the call to FreeSWITCH, which looks up in its dial
> > > plan that this number should be routed to extension (or registered
> > > user) 1000. There are no CDRs generated for either the SIP
> > > provider's gateway or the registered user in this case.
> > >
> > > In both these scenarios, if a call is inbound to FreeSWITCH but  
> has
> > > not been generated by a registered user, mod_cdr_csv does not  
> create
> > > CDRs. I have even tried to set the following parameter in
> > > cdr_csv.conf.xml      but it didn't help.
> > >
> > > <param name="legs" value="ab"/>
> > >
> > > My question is this: can mod_cdr_csv generate CDRs for inbound  
> calls
> > > from sources other than registered users? If it can't, how can  
> this
> > > be achieved?
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
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> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
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> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From david.ponzone at gmail.com  Wed May 26 22:54:36 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Thu, 27 May 2010 07:54:36 +0200
Subject: [Freeswitch-users] FS as Media Gateway Only
In-Reply-To: <AANLkTimgKRtL06xmyjXUPnpNIjlYkHC6BytzmTqKmVm7@mail.gmail.com>
References: <AANLkTimgKRtL06xmyjXUPnpNIjlYkHC6BytzmTqKmVm7@mail.gmail.com>
Message-ID: <1C47BA34-F3E5-4967-B65C-E3452C6274D1@gmail.com>

I don't think that's possible at the moment.
You can still deploy several boxes which will do SIP+RTP but I agree  
this is not the same thing.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 27/05/2010 ? 04:01, Code Ghar a ?crit :

> Is it possible -- and are there any case studies, practical  
> experience, etc -- on deploying FreeSWITCH (FS) in this  
> architecture: one server (FSSIP) handles SIP signaling only, and  
> multiple servers (FSRTP1, FSRTP2, ..., FSRTPn) handle all media  
> responsibilities? So when a call comes in, the SDP contains IP of,  
> say FSRTP1, as media handler. For this to work, FSSIP would request  
> FSRTPx for media resources for each new call and add its IP and port  
> in SDP. The media servers/gateways would play IVR, etc.; collect  
> DTMF and forward as appropriate to FSSIP; perform transcoding; etc.;  
> all while FSSIP only deals with signaling. This way multiple servers  
> could be deployed to handle media responsibilities and only a  
> handful would be required for signaling. In future if there's a  
> greater need for transcoding, etc. all you need to do is deploy a  
> media server and not have to add servers for signaling.
>
> This idea came to me because I have come across two proprietary  
> applications that do it this way. They have a SIP component and a  
> media component. You can run both on the same physical machine or  
> you can separate them out into multiple machines.
>
> Another way for this could be to integrate FS as a media component  
> to another application's SIP component. A mix-and-match, so to speak.
>
> On the flip side, deploy FS as a SIP server and use media  
> capabilities of some other hardware or software application. For  
> example, FS handles signaling and use dedicated hardware for media.  
> A good example of this is illustrated (somewhat) by an image on  
> Sangoma's website: http://www.sangoma.com/assets/images/content/transcoding_diagram.jpg 
> . Look at the "pooled transcoding".
>
> Is FS even designed to be this modular? If so, how can the  
> aforementioned scenario(s) be achieved?
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From babak.freeswitch at gmail.com  Wed May 26 23:01:12 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Thu, 27 May 2010 10:31:12 +0430
Subject: [Freeswitch-users] nat problem!
Message-ID: <AANLkTin2PsoyY9jqaq2CW6EY-IfyyUxSUQFPnHBGaQuA@mail.gmail.com>

Hi all
I googled the problem(RECOVERY_ON_TIMER_EXPIRE) and it seems it is related
to NAT, but how can I fix it? I've changed:
<param name="enable-timer" value="false"/>

but it is not working
thanx
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From ranjtech at gmail.com  Wed May 26 23:07:34 2010
From: ranjtech at gmail.com (RR)
Date: Thu, 27 May 2010 02:07:34 -0400
Subject: [Freeswitch-users] Direct inward dialling
In-Reply-To: <BE0E42CB-9888-438F-BBD7-F2CC76FAF7E0@jerris.com>
References: <491B305B.4060307@gmx.net>
	<072A175C-9004-4C14-90EC-9A93A8453787@gmail.com>
	<AANLkTinxYIG6S30XMB5GUVuWsXlyhG_slqc_e0PQy0wM@mail.gmail.com>
	<AANLkTinjUuCyrK6eVSalV6fUVNFM-j_zlvrzRxq3lJZQ@mail.gmail.com>
	<AANLkTik0JvNDXax6Ay99hrcgzuiItY0f4LkCsaRjYS8a@mail.gmail.com>
	<AANLkTinYufK3FWtbM95088fOOOWWzOyMpU6ho0VXiOEy@mail.gmail.com>
	<AANLkTinLTWiKrvgPMiiEZArMxMQFCgtJNXoHO_BBlHRN@mail.gmail.com>
	<AANLkTilq_gLLPa-VaRlaiKofqNVlpvm1zgSiL53UTf7z@mail.gmail.com>
	<AANLkTikiraunYdhjcCUmKLmgNAyQ-Sp6r4EpohHuEbZn@mail.gmail.com>
	<AANLkTimMkeVcO7mqzCVA95PE_XSGoGdHfC2A7HJ5i0_R@mail.gmail.com>
	<AANLkTim06fDEoeHe_hVVtfMDUeS2saTi-JyKiJQFB9ZC@mail.gmail.com>
	<35FA7A4D-4937-4EA1-8FA4-B124129C92DD@freeswitch.org>
	<BE0E42CB-9888-438F-BBD7-F2CC76FAF7E0@jerris.com>
Message-ID: <AANLkTimV-P_o-tE8JI6-JE27wh-vlNJGRTpvTQdLlajQ@mail.gmail.com>

hi Mike,

no these are calls coming into our network as a result of people calling the
DIDs we own and they all terminate into the FS serving as a call
distributor. Since a lot of these DIDs are International, we get the numbers
come to us in all sorts of weird formats so I needed to normalise them all
by removing all sorts of weird stuff and prefixes from them

Cheers
RR

On Wed, May 26, 2010 at 5:41 PM, Michael Jerris <mike at jerris.com> wrote:

> are these calls to a registering gateway?
>
> On May 26, 2010, at 1:40 AM, Michael S Collins wrote:
>
> It's all good. Now you have to pay it forward. :)
> -MC
>
> Sent from my iPhone
>
> On May 25, 2010, at 9:34 PM, RR <ranjtech at gmail.com> wrote:
>
> Michael,
>
> Thank you SO SO much for the help. Your regex work perfectly as desired. I
> had tried what you suggested earlier but I think I might've made a mistake
> somewhere because I wasn't getting the right results so I resorted to doing
> the "|" between the prefixes to strip them out thinking maybe FS works by
> going if it begins + OR +1 OR 011 then remove them but I guess it doesn't as
> when "\" appears it uses / matches against only the first one of the those
> as opposed to all of those.
>
> Thanks again and sorry for wasting your time ;)
>
> Cheers
> RR
>
> On Tue, May 25, 2010 at 9:19 PM, Michael Collins < <msc at freeswitch.org>
> msc at freeswitch.org> wrote:
>
>>
>>
>> On Tue, May 25, 2010 at 5:44 PM, RR < <ranjtech at gmail.com>
>> ranjtech at gmail.com> wrote:
>>
>>> Michael, haha, yeah they indeed are. That's why I'm routing based on $2,
>>> but I still see the 1 and/or the 011 going through to the "bridge"
>>> application. Why??
>>>
>> Because your regex is wrong. :) It took me a while to figure it out. I'm
>> surprised it worked at all. All the stuff you have inside the first set of
>> parens is not behaving the way you think it should be. If I read your
>> intentions correctly you're trying to strip off leading:
>> +
>> OR
>> +1
>> OR
>> 1
>>
>> In the first regex. Correct? If ANI is NANPA-ish then try this in your
>> first regex:
>> ^\+?1?([2-9]\d+).*$
>>
>> That should strip off leading + and/or 1 and capture just the 10-digit
>> phone number in $1. (Be sure to use $1 and not $2, unless you had your heart
>> set on using $2 in which case wrap the first part of the regex in parens)
>>
>> The other regex is also tricky. I assume you are trying to strip off the
>> same as above as well as 011? Try this:
>> ^\+?1?(011)?([2-9]\d+).*$
>>
>> Again, if the phone number in question is NANPA then $2 should contain
>> just the 10 digits you want. Play around with that and let us know what
>> happens. Also, don't forget what I said about using regex from the fs_cli.
>> You can test all this stuff yourself. :)
>>
>> -MC
>>
>>
>>
>>
>>>
>>> On Tue, May 25, 2010 at 8:34 PM, Michael Collins < <msc at freeswitch.org>
>>> msc at freeswitch.org> wrote:
>>>
>>>>
>>>>
>>>> On Tue, May 25, 2010 at 5:27 PM, RR < <ranjtech at gmail.com>
>>>> ranjtech at gmail.com> wrote:
>>>>
>>>>> Ok, so I take that back. This seems to only work when the dialplan has
>>>>> a specific ANI and DNIS / destination_number / sip_to_user defined. If this
>>>>> is more general
>>>>>
>>>>> like
>>>>>
>>>>> <include>
>>>>>   <extension name="public_did">
>>>>>     <condition field="ani" expression="^(\+?|\+1?|1?)(\d+).*$"
>>>>> break="never">
>>>>>         <action application="set"
>>>>> data="effective_caller_id_number=$2"/>
>>>>>         <action application="set" data="effective_caller_id_name=$2"/>
>>>>>     </condition>
>>>>>     <condition field="${sip_to_user}"
>>>>> expression="^(\+1?|\+|1?|011?)(\d+).*$" break="never">
>>>>>       <action application="set" data="continue_on_fail=false"/>
>>>>>       <action application="set" data="hangup_after_bridge=true"/>
>>>>>       <action application="set" data="domain_name=$${domain}"/>
>>>>>       <action application="set" data="bypass_media=true"/>
>>>>>       <action application="limit_hash" data="in cc_blades 4200
>>>>> !USER_BUSY"/>
>>>>>       <action application="bridge"
>>>>> data="{sip_invite_domain=${sip_from_host}}sofia/gateway/${distributor(cc_blades)}/$2"/>
>>>>>     </condition>
>>>>>   </extension>
>>>>> </include>
>>>>>
>>>>> then even though the expression/conditions seem to match, none of the
>>>>> digits are being stripped off. Shouldn't this be stripping off digits??
>>>>>
>>>>> Here's the debug output:
>>>>>
>>>>> Dialplan: sofia/external/16469NNNNNN Regex (PASS) [public_did]
>>>>> ani(16469NNNNNN;phone-context=+1) =~ /^(\+?|\+1?|1?)(\d+).*$/ break=never
>>>>> Dialplan: sofia/external/16469NNNNNN Action
>>>>> set(effective_caller_id_number=16469NNNNNN)
>>>>> Dialplan: sofia/external/16469NNNNNN Action
>>>>> set(effective_caller_id_name=16469NNNNNN)
>>>>> Dialplan: sofia/external/16469NNNNNN Regex (PASS) [public_did]
>>>>> ${sip_to_user}(011390NNNNNNNNNN;phone-context=+39) =~
>>>>> /^(\+1?|\+|1?|011?)(\d+).*$/ break=never
>>>>> Dialplan: sofia/external/16469NNNNNN Action set(continue_on_fail=false)
>>>>> Dialplan: sofia/external/16469NNNNNN Action
>>>>> set(hangup_after_bridge=true)
>>>>> Dialplan: sofia/external/16469NNNNNN Action
>>>>> set(domain_name=208.72.186.166)
>>>>> Dialplan: sofia/external/16469NNNNNN Action set(bypass_media=true)
>>>>> Dialplan: sofia/external/16469NNNNNN Action limit_hash(in cc_blades
>>>>> 4200 !USER_BUSY)
>>>>> Dialplan: sofia/external/16469NNNNNN Action
>>>>> bridge({sip_invite_domain=${sip_from_host}}sofia/gateway/${distributor(cc_blades)}/011390NNNNNNNNNN)
>>>>>
>>>>> why're the '1' in the ANI and '011' in the DNIS/sip_to_user being
>>>>> stripped off???
>>>>>
>>>>
>>>> Regex 101 :)
>>>>
>>>> The 1 or the 011 are in $1
>>>> -MC
>>>>
>>>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
>
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From vetali100 at gmail.com  Wed May 26 23:15:14 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Thu, 27 May 2010 09:15:14 +0300
Subject: [Freeswitch-users] FS as Media Gateway Only
In-Reply-To: <AANLkTimgKRtL06xmyjXUPnpNIjlYkHC6BytzmTqKmVm7@mail.gmail.com>
References: <AANLkTimgKRtL06xmyjXUPnpNIjlYkHC6BytzmTqKmVm7@mail.gmail.com>
Message-ID: <AANLkTikoxsXFuYBxsXXtB26Y9_8f1I1wGmE-u-I_l9PP@mail.gmail.com>

Hi Code,

I have working example doing exactly what you've described.
One signalling FS bridges incoming call to a set of media servers (depending
on ip, but you can implement any routing logic including round robin) and
then transfers media stream after bridging to that media server.

You can achieve this on signalling FS by creating a Lua script that contains
the following lines:

media_server="my_media_X.mydomain.com"; --to be determined by routing logic
forwarding_session = "sofia/external/"..called_number.."@"..media_server;
session:setVariable("bypass_media_after_bridge", "true");
session:setVariable("hangup_after_bridge", "true");
session:execute("bridge",forwarding_session);

The call will arrive to the selected media server successfully and media
stream will start bypassing signalling FS after bridge.

You can read the following thread, it describes how you can setup such
configuration.
http://lists.freeswitch.org/pipermail/freeswitch-users/2010-March/055231.html

I think it will fit your needs.

Regards,
Vitalie


2010/5/27 Code Ghar <codeghar at gmail.com>

> Is it possible -- and are there any case studies, practical experience, etc
> -- on deploying FreeSWITCH (FS) in this architecture: one server (FSSIP)
> handles SIP signaling only, and multiple servers (FSRTP1, FSRTP2, ...,
> FSRTPn) handle all media responsibilities? So when a call comes in, the SDP
> contains IP of, say FSRTP1, as media handler. For this to work, FSSIP would
> request FSRTPx for media resources for each new call and add its IP and port
> in SDP. The media servers/gateways would play IVR, etc.; collect DTMF and
> forward as appropriate to FSSIP; perform transcoding; etc.; all while FSSIP
> only deals with signaling. This way multiple servers could be deployed to
> handle media responsibilities and only a handful would be required for
> signaling. In future if there's a greater need for transcoding, etc. all you
> need to do is deploy a media server and not have to add servers for
> signaling.
>
> This idea came to me because I have come across two proprietary
> applications that do it this way. They have a SIP component and a media
> component. You can run both on the same physical machine or you can separate
> them out into multiple machines.
>
> Another way for this could be to integrate FS as a media component to
> another application's SIP component. A mix-and-match, so to speak.
>
> On the flip side, deploy FS as a SIP server and use media capabilities of
> some other hardware or software application. For example, FS handles
> signaling and use dedicated hardware for media. A good example of this is
> illustrated (somewhat) by an image on Sangoma's website:
> http://www.sangoma.com/assets/images/content/transcoding_diagram.jpg. Look
> at the "pooled transcoding".
>
> Is FS even designed to be this modular? If so, how can the aforementioned
> scenario(s) be achieved?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sid at eltc.ru  Wed May 26 23:23:04 2010
From: sid at eltc.ru (Sergey Scheglov)
Date: Thu, 27 May 2010 13:23:04 +0700
Subject: [Freeswitch-users] Don't work playback after bypass media mode.
In-Reply-To: <C7E0DD6C-0EBB-4C1F-A47A-D3685DF5C6FC@jerris.com>
References: <1427456515.20100526174916@eltc.ru>
	<C7E0DD6C-0EBB-4C1F-A47A-D3685DF5C6FC@jerris.com>
Message-ID: <1751544061.20100527132304@eltc.ru>

Hi, Michael.

You wrote 27 may 2010 ?., 12:30:25:

> we can't tell why because you omitted the part of the log that may  
> show why.  also, why do you set continue on fail twice in a row?

> On May 26, 2010, at 6:49 AM, Sergey Scheglov <sid at eltc.ru> wrote:
Thanks for reply,

"continue_on_fail=true" I removed (my fault)
and altered bypass_media (my reasons)
Now dialplan is
   <extension name="Local_Extension">
      <condition field="destination_number" expression="^([1-9]\d{4})$">
        <action application="set" data="dialed_extension=$1"/>
        <action application="export" data="dialed_extension=$1"/>
        <action application="set" data="ringback=${ru-ring}"/>
        <action application="set" data="bypass_media=${cond(${profile_index} == 1 ? true : false)}"/>
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set" data="continue_on_fail=SUBSCRIBER_ABSENT,USER_NOT_REGISTERED"/>
        <action application="bridge" data="user/${dialed_extension}@${domain_name}"/>
        <action application="pre_answer"/>
        <action application="sleep" data="1000"/>
        <action application="playback" data="elight/${originate_disposition}.wav"/>
      </condition>
    </extension>

Full call log here http://pastebin.freeswitch.org/13059
Also, in internal profile exist:
 inbound-late-negotiation true
 disable-transcoding true
and I used xml_curl (for directory).
-- 
? ?????????,
?????? ??????
ISP Good Line
e-mail: sid at eltc.ru
icq: 295481909
???. +79234872136




From david.ponzone at gmail.com  Wed May 26 23:42:25 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Thu, 27 May 2010 08:42:25 +0200
Subject: [Freeswitch-users] FS as Media Gateway Only
In-Reply-To: <AANLkTikoxsXFuYBxsXXtB26Y9_8f1I1wGmE-u-I_l9PP@mail.gmail.com>
References: <AANLkTimgKRtL06xmyjXUPnpNIjlYkHC6BytzmTqKmVm7@mail.gmail.com>
	<AANLkTikoxsXFuYBxsXXtB26Y9_8f1I1wGmE-u-I_l9PP@mail.gmail.com>
Message-ID: <E0BAE073-36B5-4007-A0F7-430ED332267E@gmail.com>

Vitali, Code,

I apologize, I answered too quickly.
That's actually a very smart way to do it when you don't have/want a  
proprietary protocol.

It should be possible to distribute the calls evenly by using mod_limit.
You can even take the transcoding into account.
If you know your media servers will only use G711 to the outside, you  
can call mod_limit once if the inbound call is G711 also, but you may  
call mod_limit twice or thrice or more (to be calculated) if the  
inbound call is G729.
This way, your mod_limit figures per media server will reflect the  
actual load on the server, not only the number of current calls.


David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 27/05/2010 ? 08:15, Vitalii Colosov a ?crit :

> Hi Code,
>
> I have working example doing exactly what you've described.
> One signalling FS bridges incoming call to a set of media servers  
> (depending on ip, but you can implement any routing logic including  
> round robin) and then transfers media stream after bridging to that  
> media server.
>
> You can achieve this on signalling FS by creating a Lua script that  
> contains the following lines:
>
> media_server="my_media_X.mydomain.com"; --to be determined by  
> routing logic
> forwarding_session = "sofia/ 
> external/"..called_number.."@"..media_server;
> session:setVariable("bypass_media_after_bridge", "true");
> session:setVariable("hangup_after_bridge", "true");
> session:execute("bridge",forwarding_session);
>
> The call will arrive to the selected media server successfully and  
> media stream will start bypassing signalling FS after bridge.
>
> You can read the following thread, it describes how you can setup  
> such configuration.
> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-March/055231.html
>
> I think it will fit your needs.
>
> Regards,
> Vitalie
>
>
> 2010/5/27 Code Ghar <codeghar at gmail.com>
> Is it possible -- and are there any case studies, practical  
> experience, etc -- on deploying FreeSWITCH (FS) in this  
> architecture: one server (FSSIP) handles SIP signaling only, and  
> multiple servers (FSRTP1, FSRTP2, ..., FSRTPn) handle all media  
> responsibilities? So when a call comes in, the SDP contains IP of,  
> say FSRTP1, as media handler. For this to work, FSSIP would request  
> FSRTPx for media resources for each new call and add its IP and port  
> in SDP. The media servers/gateways would play IVR, etc.; collect  
> DTMF and forward as appropriate to FSSIP; perform transcoding; etc.;  
> all while FSSIP only deals with signaling. This way multiple servers  
> could be deployed to handle media responsibilities and only a  
> handful would be required for signaling. In future if there's a  
> greater need for transcoding, etc. all you need to do is deploy a  
> media server and not have to add servers for signaling.
>
> This idea came to me because I have come across two proprietary  
> applications that do it this way. They have a SIP component and a  
> media component. You can run both on the same physical machine or  
> you can separate them out into multiple machines.
>
> Another way for this could be to integrate FS as a media component  
> to another application's SIP component. A mix-and-match, so to speak.
>
> On the flip side, deploy FS as a SIP server and use media  
> capabilities of some other hardware or software application. For  
> example, FS handles signaling and use dedicated hardware for media.  
> A good example of this is illustrated (somewhat) by an image on  
> Sangoma's website: http://www.sangoma.com/assets/images/content/transcoding_diagram.jpg 
> . Look at the "pooled transcoding".
>
> Is FS even designed to be this modular? If so, how can the  
> aforementioned scenario(s) be achieved?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From msc at freeswitch.org  Wed May 26 23:43:51 2010
From: msc at freeswitch.org (Michael S Collins)
Date: Wed, 26 May 2010 23:43:51 -0700
Subject: [Freeswitch-users] nat problem!
In-Reply-To: <AANLkTin2PsoyY9jqaq2CW6EY-IfyyUxSUQFPnHBGaQuA@mail.gmail.com>
References: <AANLkTin2PsoyY9jqaq2CW6EY-IfyyUxSUQFPnHBGaQuA@mail.gmail.com>
Message-ID: <880247BE-772D-4595-B591-D0577D9F9756@freeswitch.org>

What device device is performing the NAT? Does it have a SIP ALG you  
can turn off? See "ALG" on the wiki.

Sent from my iPhone

On May 26, 2010, at 11:01 PM, babak yakhchali <babak.freeswitch at gmail.com 
 > wrote:

> Hi all
> I googled the problem(RECOVERY_ON_TIMER_EXPIRE) and it seems it is  
> related to NAT, but how can I fix it? I've changed:
> <param name="enable-timer" value="false"/>
>
> but it is not working
> thanx
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From babak.freeswitch at gmail.com  Thu May 27 00:06:38 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Thu, 27 May 2010 11:36:38 +0430
Subject: [Freeswitch-users] nat problem!
In-Reply-To: <880247BE-772D-4595-B591-D0577D9F9756@freeswitch.org>
References: <AANLkTin2PsoyY9jqaq2CW6EY-IfyyUxSUQFPnHBGaQuA@mail.gmail.com>
	<880247BE-772D-4595-B591-D0577D9F9756@freeswitch.org>
Message-ID: <AANLkTinthmE7uLkV4g_zcuWYiyBQ4wiDf8zdXaUnpW5M@mail.gmail.com>

traffic from my ipphones to freeswitch is not passing through nat, cause
there are just some vlans. ipphones and freeswitch are not in the same vlan,
is the problem related to this structure?or is the problem just caused by
nat problems??
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From Prometheus001 at gmx.net  Thu May 27 00:45:16 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Thu, 27 May 2010 09:45:16 +0200
Subject: [Freeswitch-users] Any Idea how to let the phone ring with a
	different ring tone
Message-ID: <4BFE230C.6030802@gmx.net>

Hello,

for emergency issues I would like to let a phone ring with a different
ringer tone so that anybody can hear by the ringer tone that this call
is an alarm call.  I know that for this purpose an "Alert-Info" can be
added to the SIP header. But how has this to be invoked in FS,
especially when I want make a group call to a number of phones?
Anybody has an idea how to solve this?

Best regards
Peter


From babak.freeswitch at gmail.com  Thu May 27 02:58:18 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Thu, 27 May 2010 14:28:18 +0430
Subject: [Freeswitch-users] nat problem!
In-Reply-To: <AANLkTinthmE7uLkV4g_zcuWYiyBQ4wiDf8zdXaUnpW5M@mail.gmail.com>
References: <AANLkTin2PsoyY9jqaq2CW6EY-IfyyUxSUQFPnHBGaQuA@mail.gmail.com>
	<880247BE-772D-4595-B591-D0577D9F9756@freeswitch.org>
	<AANLkTinthmE7uLkV4g_zcuWYiyBQ4wiDf8zdXaUnpW5M@mail.gmail.com>
Message-ID: <AANLkTimG25bRUAlc8KwB8v9XjQM7xqxbnnT5OnE7mC34@mail.gmail.com>

yes you are right my ip phones are behind nat is there anyway to solve this?
I checked the packet captures and found out that only the udp port is
changed from 5060 to 49761. I'm using cisco 7941 ip phones.
thanx
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From david.ponzone at gmail.com  Thu May 27 03:19:57 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Thu, 27 May 2010 12:19:57 +0200
Subject: [Freeswitch-users] nat problem!
In-Reply-To: <AANLkTimG25bRUAlc8KwB8v9XjQM7xqxbnnT5OnE7mC34@mail.gmail.com>
References: <AANLkTin2PsoyY9jqaq2CW6EY-IfyyUxSUQFPnHBGaQuA@mail.gmail.com>
	<880247BE-772D-4595-B591-D0577D9F9756@freeswitch.org>
	<AANLkTinthmE7uLkV4g_zcuWYiyBQ4wiDf8zdXaUnpW5M@mail.gmail.com>
	<AANLkTimG25bRUAlc8KwB8v9XjQM7xqxbnnT5OnE7mC34@mail.gmail.com>
Message-ID: <F1C6AB5C-545A-487F-B660-4168A1E16219@gmail.com>

May you check if the Cisco sets the rport flag in the Via field of the  
INVITE ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 27/05/2010 ? 11:58, babak yakhchali a ?crit :

> yes you are right my ip phones are behind nat is there anyway to  
> solve this? I checked the packet captures and found out that only  
> the udp port is changed from 5060 to 49761. I'm using cisco 7941 ip  
> phones.
> thanx
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jingwei.yang at gmail.com  Thu May 27 03:35:25 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Thu, 27 May 2010 18:35:25 +0800
Subject: [Freeswitch-users]  FSComm linker issue
Message-ID: <AANLkTiliaONCmhMhVsgiXi05v5dIwASq7Zb2t-QJf1YX@mail.gmail.com>

Hello,

I encountered six linker errors when compiling the latest codes of FSComm in
VS2008. I followed the steps from here:
http://wiki.freeswitch.org/wiki/FSComm#Windows. Someone else seemed to have
the same issue but got no answers:
http://article.gmane.org/gmane.comp.telephony.freeswitch.user/21757/match=fscomm
.

Please enlighten me how to deal with these errors.

Thanks and best regards,
-Jingwei

10>fshost.obj : error LNK2019: unresolved external symbol "public:
__thiscall Channel::Channel(class QString)" (??0Channel@@QAE at VQString@@@Z)
referenced in function "private: void __thiscall
FSHost::eventChannelCreate(class QSharedPointer<struct switch_event>,class
QString)" (?eventChannelCreate at FSHost@@AAEXV?$QSharedPointer at Uswitch_event
@@@@VQString@@@Z)

10>mainwindow.obj : error LNK2019: unresolved external symbol "public:
__thiscall StateDebugDialog::StateDebugDialog(class QWidget *)"
(??0StateDebugDialog@@QAE at PAVQWidget@@@Z) referenced in function "private:
void __thiscall MainWindow::debugEventsTriggered(void)"
(?debugEventsTriggered at MainWindow@@AAEXXZ)

10>mainwindow.obj : error LNK2019: unresolved external symbol "public:
__thiscall ConsoleWindow::ConsoleWindow(class QWidget *)" (??0ConsoleWindow@
@QAE at PAVQWidget@@@Z) referenced in function "private: void __thiscall
MainWindow::debugConsoleTriggered(void)" (?debugConsoleTriggered at MainWindow
@@AAEXXZ)

10>prefdialog.obj : error LNK2019: unresolved external symbol "public:
__thiscall CodecWidget::CodecWidget(class QWidget *)" (??0CodecWidget@
@QAE at PAVQWidget@@@Z) referenced in function "public: void __thiscall
Ui_PrefDialog::setupUi(class QDialog *)" (?setupUi at Ui_PrefDialog
@@QAEXPAVQDialog@@@Z)

10>prefsofia.obj : error LNK2019: unresolved external symbol "public: void
__thiscall CodecWidget::setCodecString(class QString)"
(?setCodecString at CodecWidget@@QAEXVQString@@@Z) referenced in function
"public: void __thiscall PrefSofia::readConfig(void)" (?readConfig at PrefSofia
@@QAEXXZ)

10>prefsofia.obj : error LNK2019: unresolved external symbol "public: class
QString __thiscall CodecWidget::getCodecString(void)"
(?getCodecString at CodecWidget@@QAE?AVQString@@XZ) referenced in function
"public: void __thiscall PrefSofia::writeConfig(void)"
(?writeConfig at PrefSofia@@QAEXXZ)
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From emilbergg at gmail.com  Thu May 27 04:02:57 2010
From: emilbergg at gmail.com (Emil Berg)
Date: Thu, 27 May 2010 14:02:57 +0300
Subject: [Freeswitch-users] Disabling voice mail
In-Reply-To: <4BFD8671.6020001@gmail.com>
References: <AANLkTil6J0PvUS4rZx2lBNbIhWIDELH5k9euUZ06J0Gq@mail.gmail.com>
	<4C7FF608-66F3-4DAF-9C6F-F26C0D7FEB36@freeswitch.org>
	<4BFD8671.6020001@gmail.com>
Message-ID: <AANLkTineo6LQcVZf7g5XqLT-_KXA1ooRuN5OnEh4_pcR@mail.gmail.com>

thank u,
but it is still not working, i even tried:
         <action application="set" data="continue_on_fail=NO_ANSWER"/>
when i make a call, and the remote side is connected to the server but
doesn't answer it sends the local user OK msg and starts voice mail.
the remote user is sent a cancel msg.
i want that the cancel msg will be sent to both users.


Thank you,
Emil.


On Wed, May 26, 2010 at 11:37 PM, leo.zibi at gmail.com <leo.zibi at gmail.com>wrote:

>  or  <extension name="Local_Extension">
>
> <action application="set" data="continue_on_fail=*true*"/>
>
> change
>
> <action application="set" data="continue_on_fail=*false*"/>
>
> and reloadxml
>
>
>
> Brian West wrote:
>
> Just remove it from the dialplan...
>
> /b
>
> On May 26, 2010, at 8:19 AM, Emil Berg wrote:
>
>
>
>  Hello,
>
> I am a newbie and I'd like to disable the voice mail.
> All I want is that when calling to 1000-1010 users, the voice mail will not answer in case that the user is not available. Instead, the server will send a standard reject.
> Is there any simple way to make it through configuration?
>
> I was trying to change the dialplans, but it's very confusing and it didn't work.
>
> Thank you,
> Emil.
>
>
>  _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From frank at impactfax.com  Thu May 27 04:05:20 2010
From: frank at impactfax.com (Frank @ Impact)
Date: Thu, 27 May 2010 07:05:20 -0400
Subject: [Freeswitch-users] sip_cid_type settings
In-Reply-To: <AANLkTimDN131YUiEYny9e5MgVNgLeIZB5wDAQKV3GgsP@mail.gmail.com>
Message-ID: <BF07540547A74B1889D878A58B6EFD12@ws4>

Anthoney, 
Maybe you replied to the wrong thread or I am missing the connection
between sip_cid_type and dtmf-type.  Is there a connection?   However, I
also found that same example in about the same 10 seconds of time.  I
just was unsure how it related to sip_cid_type.  Is there a connection?
 
David,
The accountcode is a nice idea.  But I do not see it in the sample
sip_profiles as an available option to set.  If that variable is set in
the sip_profile, will it carry through and be available in the dialplan?
It seems that at the time of the dialplan, FS has not yet determined
which route it is going to take and so does not yet know which
sip_profile is going to be used and does not yet have those variables
available to the call.  Am I understanding this wrong?
 
-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: Tuesday, May 25, 2010 11:01 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] sip_cid_type settings
 
well i was not sure so I looked in the example config and searched for
dtmf
in under 10 seconds I was able to do the following from the build root.
 
cat conf/sip_profiles/internal.xml | grep dtmf
    <param name="dtmf-duration" value="2000"/>
    <!--<param name="dtmf-type" value="info"/>-->
 
 
 
 
2010/5/25 David Ponzone <david.ponzone at gmail.com>
Frank,
 
as far as I know, there is not.
You can probably simulate that by using a custom variable in the gateway
profile, like accountcode, and then matching that at the beginning of
your dialplan to set the sip_cid_type accordingly.
 
David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34
 
Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com
 
Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
l'intention exclusive de ses destinataires. Toute utilisation ou
diffusion non autoris?e est interdite. Tout message ?lectronique est
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de
ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
destinataire de ce message, merci de le d?truire imm?diatement et
d'avertir l'exp?diteur.
 
 
 
 
Le 25/05/2010 ? 12:13, Frank @ Impact a ?crit :



I can only find documentation on setting this value at the bridge level.
 
Is there a way to control this setting in the gateway profile instead of
having to do it in the bridge setup?
 
 
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
 

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



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900
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From gerrit308 at gmail.com  Thu May 27 04:13:30 2010
From: gerrit308 at gmail.com (humbr)
Date: Thu, 27 May 2010 04:13:30 -0700 (PDT)
Subject: [Freeswitch-users] Searching mailing lists
In-Reply-To: <C854AED1-27A2-4540-82A0-F7ED71B5DED5@freeswitch.org>
References: <1229079085.100.1274717262156.JavaMail.root@srvr12.remotelinkml.com>
	<0A8EE721-1247-4E78-9B99-BEFB156799A6@freeswitch.org>
	<1274910571528-5105924.post@n2.nabble.com>
	<AANLkTimTQPnFWDGTvxpfaJI8rhfpNfQR5ZINF2Jn3_gn@mail.gmail.com>
	<C854AED1-27A2-4540-82A0-F7ED71B5DED5@freeswitch.org>
Message-ID: <1274958810929-5108065.post@n2.nabble.com>


Hi Brian,
If it is just about searching then yes, Google is great. But sometimes you
want some pre-bedtime reading or just want to poke around. In which case
Nabble or the other suggestion are better.

Gerrit
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Searching-mailing-lists-tp5095136p5108065.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From david.ponzone at gmail.com  Thu May 27 04:40:18 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Thu, 27 May 2010 13:40:18 +0200
Subject: [Freeswitch-users] sip_cid_type settings
In-Reply-To: <BF07540547A74B1889D878A58B6EFD12@ws4>
References: <BF07540547A74B1889D878A58B6EFD12@ws4>
Message-ID: <00B6D9EC-5FF0-4F64-B6AE-B2228C786325@gmail.com>

Frank,

you are totally right.
I was probably tired when I sent that :)

I think too adding sip-cid-type as a gateway param would be logical  
and nice.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 27/05/2010 ? 13:05, Frank @ Impact a ?crit :

> Anthoney,
> Maybe you replied to the wrong thread or I am missing the connection  
> betweensip_cid_type and dtmf-type.  Is there a connection?    
> However, I also found that same example in about the same 10 seconds  
> of time.  I just was unsure how it related tosip_cid_type.  Is there  
> a connection?
>
> David,
> The accountcode is a nice idea.  But I do not see it in the sample  
> sip_profiles as an available option to set.  If that variable is set  
> in the sip_profile, will it carry through and be available in the  
> dialplan?  It seems that at the time of the dialplan, FS has not yet  
> determined which route it is going to take and so does not yet know  
> which sip_profile is going to be used and does not yet have those  
> variables available to the call.  Am I understanding this wrong?
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Anthony Minessale
> Sent: Tuesday, May 25, 2010 11:01 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] sip_cid_type settings
>
> well i was not sure so I looked in the example config and searched  
> for dtmf
> in under 10 seconds I was able to do the following from the build  
> root.
>
> cat conf/sip_profiles/internal.xml | grep dtmf
>     <param name="dtmf-duration" value="2000"/>
>     <!--<param name="dtmf-type" value="info"/>-->
>
>
>
>
> 2010/5/25 David Ponzone <david.ponzone at gmail.com>
> Frank,
>
> as far as I know, there is not.
> You can probably simulate that by using a custom variable in the  
> gateway profile, like accountcode, and then matching that at the  
> beginning of your dialplan to set the sip_cid_type accordingly.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et  
> ?tablis ? l'intention exclusive de ses destinataires. Toute  
> utilisation ou diffusion non autoris?e est interdite. Tout message  
> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou  
> falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le  
> d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
>
> Le 25/05/2010 ? 12:13, Frank @ Impact a ?crit :
>
>
> I can only find documentation on setting this value at the bridge  
> level.
>
> Is there a way to control this setting in the gateway profile  
> instead of having to do it in the bridge setup?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From steveu at coppice.org  Thu May 27 05:30:26 2010
From: steveu at coppice.org (Steve Underwood)
Date: Thu, 27 May 2010 20:30:26 +0800
Subject: [Freeswitch-users] [Out of curiosity] Why not use a voice modem
 for FXO?
In-Reply-To: <1274914198334-5106158.post@n2.nabble.com>
References: <1274881028458-5103502.post@n2.nabble.com>	<6596AAE7-B489-4D4A-B661-EDE2CD351DC3@freeswitch.org>	<1274882347926-5103647.post@n2.nabble.com>	<AANLkTikeqyqds4FVor8qBZ9PZgrz9dIPATPAbr2N8TCQ@mail.gmail.com>	<1274905609840-5105604.post@n2.nabble.com>	<AANLkTimvJNT2NPp63C-IeiII-BRTti_CNoWJJjKUmZVS@mail.gmail.com>
	<1274914198334-5106158.post@n2.nabble.com>
Message-ID: <4BFE65E2.5030809@coppice.org>

On 05/27/2010 06:49 AM, GillesToo wrote:
>
> Anthony Minessale wrote:
>    
>> A FXO card works as well as it can but traditionally requires an
>> echo canceler if you want quality that will not draw complaints
>>      
> Why do FXO cards generate echo issues while ATA don't? For instance, there
> are a lot of people complaining about echo with the Linksys 3102.
>    
ATAs contain echo cancellers. The SPA3102 contains one, but it seems to 
be badly broken. The FXO cards are mostly dumb. Its up to host software 
to provide functions like DTMF decoding and echo cancellation for them. 
Try OSLEC with most FXO cards and you'll get great results.
> Anthony Minessale wrote:
>    
>> The good thing about an ATA is that it's self-contained and translates
>> everything into SIP for you when you don't have room for hardware.
>>      
> Actually, that's why I prefer to use a PCI card instead of an ATA: No need
> for a transformer and a cable to connect the ATA to the Freeswitch server,
> and only one cable left to connect the PCI card to the wall plug. With
> non-techie customers, it seems like a safer solution.
>    
Steve



From steveu at coppice.org  Thu May 27 05:35:23 2010
From: steveu at coppice.org (Steve Underwood)
Date: Thu, 27 May 2010 20:35:23 +0800
Subject: [Freeswitch-users] [Out of curiosity] Why not use a voice modem
 for FXO?
In-Reply-To: <1274905609840-5105604.post@n2.nabble.com>
References: <1274881028458-5103502.post@n2.nabble.com>	<6596AAE7-B489-4D4A-B661-EDE2CD351DC3@freeswitch.org>	<1274882347926-5103647.post@n2.nabble.com>	<AANLkTikeqyqds4FVor8qBZ9PZgrz9dIPATPAbr2N8TCQ@mail.gmail.com>
	<1274905609840-5105604.post@n2.nabble.com>
Message-ID: <4BFE670B.6050401@coppice.org>

On 05/27/2010 04:26 AM, GillesToo wrote:
>
> Anthony Minessale wrote:
>    
>> Even if you get it working the reward is not worth the effort. You would
>> experience a lot of echo and other problems
>>      
> I read somewhere that the reason for those problems people often have with
> X100P cards (such as echo or bad far-end disconnect supervision) is that
> most of them have they use the Silicon Labs DAA chips Si3012/Si3035 which is
> only good for use in countries that use the FCC standard; Countries that use
> the CTR21 standard (Europe, and others) require the Si3014/Si3034 chips,
> which support global line standards.
>    
Most X100P clones don't use any kind of Silicon Labs chip. Whatever chip 
they do use will probably not be programmed properly to match the line, 
because the drivers don't support that. However, with the use of a 
decent echo canceller you'll probably never notice. Use OSLEC with most 
X100P cards on most lines around the world and the results should be 
pretty good. The only things those X100P cards can't do, which some more 
complex FXO cards can, is detect line current breaks and reversals. Most 
lines around the world don't produce breaks or reversals so its not a 
major issue. Everything else is down to what the host software does.
> Anthony Minessale wrote:
>    
>> ... and the time you spent working on it would be better spent doing
>> something more meaningful and using the money you received to buy a real
>> TDM card or better still, a FXO->SIP ATA.
>>      
> I thought TDM cards were a better solution to connect an IP PBX to a
> landline. Why would you recommend an FXO-SIP ATA instead? And which
> brand/model of ATA is recommended for use with Freeswitch?
>    

Steve



From anthony.minessale at gmail.com  Thu May 27 06:34:15 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 May 2010 08:34:15 -0500
Subject: [Freeswitch-users] Don't work playback after bypass media mode.
In-Reply-To: <1751544061.20100527132304@eltc.ru>
References: <1427456515.20100526174916@eltc.ru>
	<C7E0DD6C-0EBB-4C1F-A47A-D3685DF5C6FC@jerris.com>
	<1751544061.20100527132304@eltc.ru>
Message-ID: <AANLkTimUWPk_SoDoIRHbkfISJz0ibNfyWLulZxsn9gQ0@mail.gmail.com>

playback executes on line 87 of your trace.

If you do not hear the audio, It means the re-establishment of media fails
somehow based on your topology or
the phone you are on does not support early media, change your pre-answer
before the playback to answer to verify.

Try adding sofia profile internal siptrace on to see the sip traffic too.




2010/5/27 Sergey Scheglov <sid at eltc.ru>

> Hi, Michael.
>
> You wrote 27 may 2010 ?., 12:30:25:
>
> > we can't tell why because you omitted the part of the log that may
> > show why.  also, why do you set continue on fail twice in a row?
>
> > On May 26, 2010, at 6:49 AM, Sergey Scheglov <sid at eltc.ru> wrote:
> Thanks for reply,
>
> "continue_on_fail=true" I removed (my fault)
> and altered bypass_media (my reasons)
> Now dialplan is
>    <extension name="Local_Extension">
>      <condition field="destination_number" expression="^([1-9]\d{4})$">
>        <action application="set" data="dialed_extension=$1"/>
>        <action application="export" data="dialed_extension=$1"/>
>        <action application="set" data="ringback=${ru-ring}"/>
>         <action application="set"
> data="bypass_media=${cond(${profile_index} == 1 ? true : false)}"/>
>         <action application="set" data="hangup_after_bridge=true"/>
>        <action application="set"
> data="continue_on_fail=SUBSCRIBER_ABSENT,USER_NOT_REGISTERED"/>
>         <action application="bridge" data="user/${dialed_extension}@
> ${domain_name}"/>
>        <action application="pre_answer"/>
>        <action application="sleep" data="1000"/>
>        <action application="playback"
> data="elight/${originate_disposition}.wav"/>
>      </condition>
>    </extension>
>
> Full call log here http://pastebin.freeswitch.org/13059
> Also, in internal profile exist:
>  inbound-late-negotiation true
>  disable-transcoding true
> and I used xml_curl (for directory).
> --
> ? ?????????,
> ?????? ??????
> ISP Good Line
> e-mail: sid at eltc.ru
> icq: 295481909
> ???. +79234872136
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <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
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From anthony.minessale at gmail.com  Thu May 27 06:37:24 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 May 2010 08:37:24 -0500
Subject: [Freeswitch-users] Disabling voice mail
In-Reply-To: <AANLkTineo6LQcVZf7g5XqLT-_KXA1ooRuN5OnEh4_pcR@mail.gmail.com>
References: <AANLkTil6J0PvUS4rZx2lBNbIhWIDELH5k9euUZ06J0Gq@mail.gmail.com>
	<4C7FF608-66F3-4DAF-9C6F-F26C0D7FEB36@freeswitch.org>
	<4BFD8671.6020001@gmail.com>
	<AANLkTineo6LQcVZf7g5XqLT-_KXA1ooRuN5OnEh4_pcR@mail.gmail.com>
Message-ID: <AANLkTilEWyCNRJiyDwVELFOs4xxT67Yd7V9KyMQu3KQa@mail.gmail.com>

Cancel can only be doing by origination side not far end.
Edit you dialplan and comment out the voicemail line if you don't want
voicemail.

If it doesn't work, keep trying, you probably did it wrong.
People will get frustrated with you if you continue to just keep trying 1
step
and saying "it still doesn't work".  Try a few things, play around, figure
it out.


On Thu, May 27, 2010 at 6:02 AM, Emil Berg <emilbergg at gmail.com> wrote:

> thank u,
> but it is still not working, i even tried:
>          <action application="set" data="continue_on_fail=NO_ANSWER"/>
> when i make a call, and the remote side is connected to the server but
> doesn't answer it sends the local user OK msg and starts voice mail.
> the remote user is sent a cancel msg.
> i want that the cancel msg will be sent to both users.
>
>
> Thank you,
> Emil.
>
>
>
> On Wed, May 26, 2010 at 11:37 PM, leo.zibi at gmail.com <leo.zibi at gmail.com>wrote:
>
>>  or  <extension name="Local_Extension">
>>
>> <action application="set" data="continue_on_fail=*true*"/>
>>
>> change
>>
>> <action application="set" data="continue_on_fail=*false*"/>
>>
>> and reloadxml
>>
>>
>>
>> Brian West wrote:
>>
>> Just remove it from the dialplan...
>>
>> /b
>>
>> On May 26, 2010, at 8:19 AM, Emil Berg wrote:
>>
>>
>>
>>  Hello,
>>
>> I am a newbie and I'd like to disable the voice mail.
>> All I want is that when calling to 1000-1010 users, the voice mail will not answer in case that the user is not available. Instead, the server will send a standard reject.
>> Is there any simple way to make it through configuration?
>>
>> I was trying to change the dialplans, but it's very confusing and it didn't work.
>>
>> Thank you,
>> Emil.
>>
>>
>>  _______________________________________________
>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
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
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From anthony.minessale at gmail.com  Thu May 27 06:39:48 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 May 2010 08:39:48 -0500
Subject: [Freeswitch-users] Any Idea how to let the phone ring with a
	different ring tone
In-Reply-To: <4BFE230C.6030802@gmx.net>
References: <4BFE230C.6030802@gmx.net>
Message-ID: <AANLkTinjGN1Nk5CP4JA6PHiGjBrbSkgEXdjskgFKrjxc@mail.gmail.com>

you add it as a variable in your dial string

{sip_h_Alert-Info=<whatever it needs to say>}sofia/profile/foo at bar.com

On Thu, May 27, 2010 at 2:45 AM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> Hello,
>
> for emergency issues I would like to let a phone ring with a different
> ringer tone so that anybody can hear by the ringer tone that this call
> is an alarm call.  I know that for this purpose an "Alert-Info" can be
> added to the SIP header. But how has this to be invoked in FS,
> especially when I want make a group call to a number of phones?
> Anybody has an idea how to solve this?
>
> Best regards
> Peter
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
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
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From anthony.minessale at gmail.com  Thu May 27 06:44:53 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 May 2010 08:44:53 -0500
Subject: [Freeswitch-users] sip_cid_type settings
In-Reply-To: <00B6D9EC-5FF0-4F64-B6AE-B2228C786325@gmail.com>
References: <BF07540547A74B1889D878A58B6EFD12@ws4>
	<00B6D9EC-5FF0-4F64-B6AE-B2228C786325@gmail.com>
Message-ID: <AANLkTikjMLzb92uv4CdOz2O977pbzmGM2IsRFE3kp05a@mail.gmail.com>

ya, wrong param,
This is the right one:

<!-- Remote-Party-ID header -->
<param name="caller-id-type" value="rpid"/>

or

<!-- P-*-Identity family of headers -->
<param name="caller-id-type" value="pid"/>

or

<param name="caller-id-type" value="none"/>




On Thu, May 27, 2010 at 6:40 AM, David Ponzone <david.ponzone at gmail.com>wrote:

> Frank,
>
> you are totally right.
> I was probably tired when I sent that :)
>
> I think too adding sip-cid-type as a gateway param would be logical and
> nice.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 27/05/2010 ? 13:05, Frank @ Impact a ?crit :
>
> Anthoney,
> Maybe you replied to the wrong thread or I am missing the connection
> betweensip_cid_type and dtmf-type.  Is there a connection?   However, I
> also found that same example in about the same 10 seconds of time.  I just
> was unsure how it related tosip_cid_type.  Is there a connection?
>
> David,
> The accountcode is a nice idea.  But I do not see it in the sample
> sip_profiles as an available option to set.  If that variable is set in
> the sip_profile, will it carry through and be available in the dialplan?  It
> seems that at the time of the dialplan, FS has not yet determined which
> route it is going to take and so does not yet know which sip_profile is
> going to be used and does not yet have those variables available to the
> call.  Am I understanding this wrong?
>
> -----Original Message-----
> *From:* freeswitch-users-bounces at lists.freeswitch.org [
> mailto:freeswitch-users-bounces at lists.freeswitch.org<freeswitch-users-bounces at lists.freeswitch.org>
> ] *On Behalf Of *Anthony Minessale
> *Sent:* Tuesday, May 25, 2010 11:01 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] sip_cid_type settings
>
> well i was not sure so I looked in the example config and searched for dtmf
> in under 10 seconds I was able to do the following from the build root.
>
> cat conf/sip_profiles/internal.xml | grep dtmf
>     <param name="dtmf-duration" value="2000"/>
>     <!--<param name="dtmf-type" value="info"/>-->
>
>
>
>
> 2010/5/25 David Ponzone <david.ponzone at gmail.com>
> Frank,
>
> as far as I know, there is not.
> You can probably simulate that by using a custom variable in the gateway
> profile, like accountcode, and then matching that at the beginning of your
> dialplan to set the sip_cid_type accordingly.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> *www.ipeva.fr*  -   *www.ipeva-studio.com*
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message
> s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de
> ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.*
> * *
>
>
>
> Le 25/05/2010 ? 12:13, Frank @ Impact a ?crit :
>
>
> I can only find documentation on setting this value at the bridge level.
>
> Is there a way to control this setting in the gateway profile instead of
> having to do it in the bridge setup?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> 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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
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
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From anthony.minessale at gmail.com  Thu May 27 06:51:49 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 May 2010 08:51:49 -0500
Subject: [Freeswitch-users] FS as Media Gateway Only
In-Reply-To: <E0BAE073-36B5-4007-A0F7-430ED332267E@gmail.com>
References: <AANLkTimgKRtL06xmyjXUPnpNIjlYkHC6BytzmTqKmVm7@mail.gmail.com>
	<AANLkTikoxsXFuYBxsXXtB26Y9_8f1I1wGmE-u-I_l9PP@mail.gmail.com>
	<E0BAE073-36B5-4007-A0F7-430ED332267E@gmail.com>
Message-ID: <AANLkTikatpyMrEcEHxm__mhMutSngAyHbZToJkeKrzvx@mail.gmail.com>

Also consider using a sip proxy to distribute the calls to the cluster of
media boxes.
This is one of the main reasons we have suffered through SIP so we could
have proxies and redirectors based on http =/


On Thu, May 27, 2010 at 1:42 AM, David Ponzone <david.ponzone at gmail.com>wrote:

> Vitali, Code,
>
> I apologize, I answered too quickly.
> That's actually a very smart way to do it when you don't have/want a
> proprietary protocol.
>
> It should be possible to distribute the calls evenly by using mod_limit.
> You can even take the transcoding into account.
> If you know your media servers will only use G711 to the outside, you can
> call mod_limit once if the inbound call is G711 also, but you may call
> mod_limit twice or thrice or more (to be calculated) if the inbound call is
> G729.
> This way, your mod_limit figures per media server will reflect the actual
> load on the server, not only the number of current calls.
>
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 27/05/2010 ? 08:15, Vitalii Colosov a ?crit :
>
> Hi Code,
>
> I have working example doing exactly what you've described.
> One signalling FS bridges incoming call to a set of media servers
> (depending on ip, but you can implement any routing logic including round
> robin) and then transfers media stream after bridging to that media server.
>
> You can achieve this on signalling FS by creating a Lua script that
> contains the following lines:
>
> media_server="my_media_X.mydomain.com"; --to be determined by routing
> logic
> forwarding_session = "sofia/external/"..called_number.."@"..media_server;
> session:setVariable("bypass_media_after_bridge", "true");
> session:setVariable("hangup_after_bridge", "true");
> session:execute("bridge",forwarding_session);
>
> The call will arrive to the selected media server successfully and media
> stream will start bypassing signalling FS after bridge.
>
> You can read the following thread, it describes how you can setup such
> configuration.
>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-March/055231.html
>
> I think it will fit your needs.
>
> Regards,
> Vitalie
>
>
> 2010/5/27 Code Ghar <codeghar at gmail.com>
>
>> Is it possible -- and are there any case studies, practical experience,
>> etc -- on deploying FreeSWITCH (FS) in this architecture: one server (FSSIP)
>> handles SIP signaling only, and multiple servers (FSRTP1, FSRTP2, ...,
>> FSRTPn) handle all media responsibilities? So when a call comes in, the SDP
>> contains IP of, say FSRTP1, as media handler. For this to work, FSSIP would
>> request FSRTPx for media resources for each new call and add its IP and port
>> in SDP. The media servers/gateways would play IVR, etc.; collect DTMF and
>> forward as appropriate to FSSIP; perform transcoding; etc.; all while FSSIP
>> only deals with signaling. This way multiple servers could be deployed to
>> handle media responsibilities and only a handful would be required for
>> signaling. In future if there's a greater need for transcoding, etc. all you
>> need to do is deploy a media server and not have to add servers for
>> signaling.
>>
>> This idea came to me because I have come across two proprietary
>> applications that do it this way. They have a SIP component and a media
>> component. You can run both on the same physical machine or you can separate
>> them out into multiple machines.
>>
>> Another way for this could be to integrate FS as a media component to
>> another application's SIP component. A mix-and-match, so to speak.
>>
>> On the flip side, deploy FS as a SIP server and use media capabilities of
>> some other hardware or software application. For example, FS handles
>> signaling and use dedicated hardware for media. A good example of this is
>> illustrated (somewhat) by an image on Sangoma's website:
>> http://www.sangoma.com/assets/images/content/transcoding_diagram.jpg.
>> Look at the "pooled transcoding".
>>
>> Is FS even designed to be this modular? If so, how can the aforementioned
>> scenario(s) be achieved?
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
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
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From babak.freeswitch at gmail.com  Thu May 27 06:51:57 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Thu, 27 May 2010 18:21:57 +0430
Subject: [Freeswitch-users] nat problem!
In-Reply-To: <F1C6AB5C-545A-487F-B660-4168A1E16219@gmail.com>
References: <AANLkTin2PsoyY9jqaq2CW6EY-IfyyUxSUQFPnHBGaQuA@mail.gmail.com>
	<880247BE-772D-4595-B591-D0577D9F9756@freeswitch.org>
	<AANLkTinthmE7uLkV4g_zcuWYiyBQ4wiDf8zdXaUnpW5M@mail.gmail.com>
	<AANLkTimG25bRUAlc8KwB8v9XjQM7xqxbnnT5OnE7mC34@mail.gmail.com>
	<F1C6AB5C-545A-487F-B660-4168A1E16219@gmail.com>
Message-ID: <AANLkTikBTfJbSaWLA_7EUMNPBYlNZbkPEKLbJF8y64Pm@mail.gmail.com>

no it is not setting rport flag
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From codecomplete at free.fr  Thu May 27 06:53:41 2010
From: codecomplete at free.fr (GillesToo)
Date: Thu, 27 May 2010 06:53:41 -0700 (PDT)
Subject: [Freeswitch-users] Searching mailing lists
In-Reply-To: <1229079085.100.1274717262156.JavaMail.root@srvr12.remotelinkml.com>
References: <1229079085.100.1274717262156.JavaMail.root@srvr12.remotelinkml.com>
Message-ID: <1274968421503-5108699.post@n2.nabble.com>



David Swardstrom wrote:
> However, there does not seem to be a search engine capability supplied. Is
> there a search capability? If so, is there a web page for this?

I second using the Nabble access to the Freeswitch mailing list. Much easier
to reply correctly when using the digest mode, and you can search the
archives easily.

http://freeswitch-users.2379917.n2.nabble.com/ 
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Searching-mailing-lists-tp5095136p5108699.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From codecomplete at free.fr  Thu May 27 07:04:16 2010
From: codecomplete at free.fr (GillesToo)
Date: Thu, 27 May 2010 07:04:16 -0700 (PDT)
Subject: [Freeswitch-users] [Out of curiosity] Why not use a voice modem
	for FXO?
In-Reply-To: <4BFE670B.6050401@coppice.org>
References: <1274881028458-5103502.post@n2.nabble.com>
	<6596AAE7-B489-4D4A-B661-EDE2CD351DC3@freeswitch.org>
	<1274882347926-5103647.post@n2.nabble.com>
	<AANLkTikeqyqds4FVor8qBZ9PZgrz9dIPATPAbr2N8TCQ@mail.gmail.com>
	<1274905609840-5105604.post@n2.nabble.com>
	<4BFE670B.6050401@coppice.org>
Message-ID: <1274969056725-5108758.post@n2.nabble.com>


Most X100P clones don't use any kind of Silicon Labs chip. Whatever chip they
do use will probably not be programmed properly to match the line, because
the drivers don't support that. However, with the use of a decent echo
canceller you'll probably never notice. Use OSLEC with most X100P cards on
most lines around the world and the results should be pretty good.

Thanks for the clarification. I have a bunch of X10xP cards, and they all
have chips that say "Si". Unfortunately, the ones that have the supposedly
Europe-compatible chip (3014 + 3021)... I couldn't get to work on the
computers I have here. I guess it's a hardware issue.


Steve Underwood wrote:
> The only things those X100P cards can't do, which some more  complex FXO
> cards can, is detect line current breaks and reversals. Most lines around
> the world don't produce breaks or reversals so its not a major issue.
> Everything else is down to what the host software does.

Too bad no one followed the X10xP project and came up with a reliable,
entry-level card for SOHO users. $100 is almost the price of the PC on which
Freeswitch is running :-)

Thanks for the technical infos.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Out-of-curiosity-Why-not-use-a-voice-modem-for-FXO-tp5103502p5108758.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From brian at freeswitch.org  Thu May 27 07:50:55 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 27 May 2010 09:50:55 -0500
Subject: [Freeswitch-users] Any Idea how to let the phone ring with a
	different ring tone
In-Reply-To: <AANLkTinjGN1Nk5CP4JA6PHiGjBrbSkgEXdjskgFKrjxc@mail.gmail.com>
References: <4BFE230C.6030802@gmx.net>
	<AANLkTinjGN1Nk5CP4JA6PHiGjBrbSkgEXdjskgFKrjxc@mail.gmail.com>
Message-ID: <C01CA28A-84A7-4717-B722-DE447B51ABB2@freeswitch.org>

This is one of the cases that doesn't follow the convention its "alert_info" as the variable name.

/b

On May 27, 2010, at 8:39 AM, Anthony Minessale wrote:

> you add it as a variable in your dial string
> 
> {sip_h_Alert-Info=<whatever it needs to say>}sofia/profile/foo at bar.com

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From dswardstrom at remotelink.com  Thu May 27 09:08:58 2010
From: dswardstrom at remotelink.com (David Swardstrom)
Date: Thu, 27 May 2010 11:08:58 -0500 (CDT)
Subject: [Freeswitch-users] libtool version mismatch
Message-ID: <1533143392.107.1274976538011.JavaMail.root@srvr12.remotelinkml.com>

The only local changes to the code were added (with modifications) by
Anthony under JIRA FSCORE-612. So I just did a rm -r * while in /usr/src/freeswitch.
Then redid the git to re-create everything.

The significant change that I made was to delete a file (or files).
I have used TortiseSVN on Windows to rebuild (by doing a full revert) the
directory structure. Have not figured out how to use git to do this.


Regards,
Paul David Swardstrom



From babak.freeswitch at gmail.com  Thu May 27 09:39:25 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Thu, 27 May 2010 21:09:25 +0430
Subject: [Freeswitch-users] nat problem!
In-Reply-To: <AANLkTikBTfJbSaWLA_7EUMNPBYlNZbkPEKLbJF8y64Pm@mail.gmail.com>
References: <AANLkTin2PsoyY9jqaq2CW6EY-IfyyUxSUQFPnHBGaQuA@mail.gmail.com>
	<880247BE-772D-4595-B591-D0577D9F9756@freeswitch.org>
	<AANLkTinthmE7uLkV4g_zcuWYiyBQ4wiDf8zdXaUnpW5M@mail.gmail.com>
	<AANLkTimG25bRUAlc8KwB8v9XjQM7xqxbnnT5OnE7mC34@mail.gmail.com>
	<F1C6AB5C-545A-487F-B660-4168A1E16219@gmail.com>
	<AANLkTikBTfJbSaWLA_7EUMNPBYlNZbkPEKLbJF8y64Pm@mail.gmail.com>
Message-ID: <AANLkTilNIDxTurbL7CGnUvMjkbdDi2kSZD2EhDaodP-q@mail.gmail.com>

finally I found out what the scenario is:
I captured packets on registration and I found out when registering,
freeswitch uses port 5060 as destination even if incoming udp packets are
from ports other than 5060.
but during the call freeswitch uses the source port of the incoming packet
and the problem arises.
Is there anyway that I can force freeswitch to always use the port specified
in via header?
thanx
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From anthony.minessale at gmail.com  Thu May 27 09:51:57 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 May 2010 11:51:57 -0500
Subject: [Freeswitch-users] nat problem!
In-Reply-To: <AANLkTilNIDxTurbL7CGnUvMjkbdDi2kSZD2EhDaodP-q@mail.gmail.com>
References: <AANLkTin2PsoyY9jqaq2CW6EY-IfyyUxSUQFPnHBGaQuA@mail.gmail.com>
	<880247BE-772D-4595-B591-D0577D9F9756@freeswitch.org>
	<AANLkTinthmE7uLkV4g_zcuWYiyBQ4wiDf8zdXaUnpW5M@mail.gmail.com>
	<AANLkTimG25bRUAlc8KwB8v9XjQM7xqxbnnT5OnE7mC34@mail.gmail.com>
	<F1C6AB5C-545A-487F-B660-4168A1E16219@gmail.com>
	<AANLkTikBTfJbSaWLA_7EUMNPBYlNZbkPEKLbJF8y64Pm@mail.gmail.com>
	<AANLkTilNIDxTurbL7CGnUvMjkbdDi2kSZD2EhDaodP-q@mail.gmail.com>
Message-ID: <AANLkTikCD7TvNAfF2dwhB95SLwwVE0A_UxO07pJQDmkK@mail.gmail.com>

yes it's called rport



On Thu, May 27, 2010 at 11:39 AM, babak yakhchali <
babak.freeswitch at gmail.com> wrote:

> finally I found out what the scenario is:
> I captured packets on registration and I found out when registering,
> freeswitch uses port 5060 as destination even if incoming udp packets are
> from ports other than 5060.
> but during the call freeswitch uses the source port of the incoming packet
> and the problem arises.
> Is there anyway that I can force freeswitch to always use the port
> specified in via header?
> thanx
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <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
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From david.ponzone at ipeva.fr  Thu May 27 09:56:00 2010
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 27 May 2010 18:56:00 +0200
Subject: [Freeswitch-users] nat problem!
In-Reply-To: <AANLkTilNIDxTurbL7CGnUvMjkbdDi2kSZD2EhDaodP-q@mail.gmail.com>
References: <AANLkTin2PsoyY9jqaq2CW6EY-IfyyUxSUQFPnHBGaQuA@mail.gmail.com>
	<880247BE-772D-4595-B591-D0577D9F9756@freeswitch.org>
	<AANLkTinthmE7uLkV4g_zcuWYiyBQ4wiDf8zdXaUnpW5M@mail.gmail.com>
	<AANLkTimG25bRUAlc8KwB8v9XjQM7xqxbnnT5OnE7mC34@mail.gmail.com>
	<F1C6AB5C-545A-487F-B660-4168A1E16219@gmail.com>
	<AANLkTikBTfJbSaWLA_7EUMNPBYlNZbkPEKLbJF8y64Pm@mail.gmail.com>
	<AANLkTilNIDxTurbL7CGnUvMjkbdDi2kSZD2EhDaodP-q@mail.gmail.com>
Message-ID: <3217BC91-AFC6-45C6-BB53-EB09E4F84E62@ipeva.fr>

force-rport

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 27/05/2010 ? 18:39, babak yakhchali a ?crit :

> finally I found out what the scenario is:
> I captured packets on registration and I found out when registering,  
> freeswitch uses port 5060 as destination even if incoming udp  
> packets are from ports other than 5060.
> but during the call freeswitch uses the source port of the incoming  
> packet and the problem arises.
> Is there anyway that I can force freeswitch to always use the port  
> specified in via header?
> thanx
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Thu May 27 09:56:08 2010
From: brian at freeswitch.org (Brian West)
Date: Thu, 27 May 2010 11:56:08 -0500
Subject: [Freeswitch-users] nat problem!
In-Reply-To: <AANLkTilNIDxTurbL7CGnUvMjkbdDi2kSZD2EhDaodP-q@mail.gmail.com>
References: <AANLkTin2PsoyY9jqaq2CW6EY-IfyyUxSUQFPnHBGaQuA@mail.gmail.com>
	<880247BE-772D-4595-B591-D0577D9F9756@freeswitch.org>
	<AANLkTinthmE7uLkV4g_zcuWYiyBQ4wiDf8zdXaUnpW5M@mail.gmail.com>
	<AANLkTimG25bRUAlc8KwB8v9XjQM7xqxbnnT5OnE7mC34@mail.gmail.com>
	<F1C6AB5C-545A-487F-B660-4168A1E16219@gmail.com>
	<AANLkTikBTfJbSaWLA_7EUMNPBYlNZbkPEKLbJF8y64Pm@mail.gmail.com>
	<AANLkTilNIDxTurbL7CGnUvMjkbdDi2kSZD2EhDaodP-q@mail.gmail.com>
Message-ID: <C9CE0B47-8479-4EEE-9C78-24A29B55D0F2@freeswitch.org>


On May 27, 2010, at 11:39 AM, babak yakhchali wrote:

> finally I found out what the scenario is:
> I captured packets on registration and I found out when registering, freeswitch uses port 5060 as destination even if incoming udp packets are from ports other than 5060.
> but during the call freeswitch uses the source port of the incoming packet and the problem arises.

That is why the client needs to enable rport.  because when they register I suspect the contact said 5060 in it.  We tend to do exactly what you tell us in the contact field unless rport is on.

> Is there anyway that I can force freeswitch to always use the port specified in via header?

NDLB-force-rport on the profile but that can break some devices.

> thanx



From david.ponzone at gmail.com  Thu May 27 09:56:09 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Thu, 27 May 2010 18:56:09 +0200
Subject: [Freeswitch-users] nat problem!
In-Reply-To: <AANLkTilNIDxTurbL7CGnUvMjkbdDi2kSZD2EhDaodP-q@mail.gmail.com>
References: <AANLkTin2PsoyY9jqaq2CW6EY-IfyyUxSUQFPnHBGaQuA@mail.gmail.com>
	<880247BE-772D-4595-B591-D0577D9F9756@freeswitch.org>
	<AANLkTinthmE7uLkV4g_zcuWYiyBQ4wiDf8zdXaUnpW5M@mail.gmail.com>
	<AANLkTimG25bRUAlc8KwB8v9XjQM7xqxbnnT5OnE7mC34@mail.gmail.com>
	<F1C6AB5C-545A-487F-B660-4168A1E16219@gmail.com>
	<AANLkTikBTfJbSaWLA_7EUMNPBYlNZbkPEKLbJF8y64Pm@mail.gmail.com>
	<AANLkTilNIDxTurbL7CGnUvMjkbdDi2kSZD2EhDaodP-q@mail.gmail.com>
Message-ID: <BEF3111D-C78F-4BF3-9C26-D066EC128A66@gmail.com>

force-rport

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 27/05/2010 ? 18:39, babak yakhchali a ?crit :

> finally I found out what the scenario is:
> I captured packets on registration and I found out when registering,  
> freeswitch uses port 5060 as destination even if incoming udp  
> packets are from ports other than 5060.
> but during the call freeswitch uses the source port of the incoming  
> packet and the problem arises.
> Is there anyway that I can force freeswitch to always use the port  
> specified in via header?
> thanx
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From sid at eltc.ru  Thu May 27 11:31:26 2010
From: sid at eltc.ru (Sergey Scheglov)
Date: Fri, 28 May 2010 01:31:26 +0700
Subject: [Freeswitch-users] Don't work playback after bypass media mode.
In-Reply-To: <AANLkTimUWPk_SoDoIRHbkfISJz0ibNfyWLulZxsn9gQ0@mail.gmail.com>
References: <1427456515.20100526174916@eltc.ru>
	<C7E0DD6C-0EBB-4C1F-A47A-D3685DF5C6FC@jerris.com>
	<1751544061.20100527132304@eltc.ru>
	<AANLkTimUWPk_SoDoIRHbkfISJz0ibNfyWLulZxsn9gQ0@mail.gmail.com>
Message-ID: <264031947.20100528013126@eltc.ru>

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From sid at eltc.ru  Thu May 27 11:35:47 2010
From: sid at eltc.ru (Sergey Scheglov)
Date: Fri, 28 May 2010 01:35:47 +0700
Subject: [Freeswitch-users] Don't work playback after bypass media mode.
In-Reply-To: <AANLkTimUWPk_SoDoIRHbkfISJz0ibNfyWLulZxsn9gQ0@mail.gmail.com>
References: <1427456515.20100526174916@eltc.ru>
	<C7E0DD6C-0EBB-4C1F-A47A-D3685DF5C6FC@jerris.com>
	<1751544061.20100527132304@eltc.ru>
	<AANLkTimUWPk_SoDoIRHbkfISJz0ibNfyWLulZxsn9gQ0@mail.gmail.com>
Message-ID: <141728530.20100528013547@eltc.ru>

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From anthony.minessale at gmail.com  Thu May 27 11:46:05 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 May 2010 13:46:05 -0500
Subject: [Freeswitch-users] Don't work playback after bypass media mode.
In-Reply-To: <141728530.20100528013547@eltc.ru>
References: <1427456515.20100526174916@eltc.ru>
	<C7E0DD6C-0EBB-4C1F-A47A-D3685DF5C6FC@jerris.com>
	<1751544061.20100527132304@eltc.ru>
	<AANLkTimUWPk_SoDoIRHbkfISJz0ibNfyWLulZxsn9gQ0@mail.gmail.com>
	<141728530.20100528013547@eltc.ru>
Message-ID: <AANLkTinpkGvmvUvOQzAQdmUHHMNN_kxWQY304wlMEBwC@mail.gmail.com>

it's against the SIP spec to re-negotiate media before you have answered.
the transaction to negotiate the SDP from the original call has not
completed so it's illegal to send a re-invite.

instead you should use bypass_media_after_bridge=true so the bypass only
happens when the bridge works.


2010/5/27 Sergey Scheglov <sid at eltc.ru>

>  Hi, Anthony.
>
>
> You wrote 27 may 2010 ?., 20:34:15:
>
>
>
>  playback executes on line 87 of your trace.
>
>
> If you do not hear the audio, It means the re-establishment of media fails
> somehow based on your topology or
>
> the phone you are on does not support early media, change your pre-answer
> before the playback to answer to verify.
>
>
> Try adding sofia profile internal siptrace on to see the sip traffic too.
>
>
>
>
> > playback executes on line 87 of your trace.
>
>
> yes, but executed by log after 30 seconds after line 84 and immediately
> hangup.
>
> Note, duration wav file - 4 sec.
>
>
> > If you do not hear the audio, It means the re-establishment of media
> fails somehow based on your topology or
>
> the phone you are on does not support early media,
>
>
> I don't hear the audio, because FS don't send RTP traffic to my phone in
> early media mode (checked sniffers) in my case.
>
> If dialplans is:
>
> <extension name="fail_balance">
>
> <condition field="${blabla}" expression="^\-1$">
>
> <action application="pre_answer"/>
>
> <action application="sleep" data="1000"/>
>
> <action application="playback" data="elight/neg_balance.wav"/>
>
> <action application="hangup" data="BEARERCAPABILITY_NOTAUTH"/>
>
> </condition>
>
> </extension>
>
> then work's fine, no problems (bypass_media=true not set).
>
>
> > change your pre-answer before the playback to answer to verify.
>
>
> If set answer, problem disappears. But answer it's not for my case.
>
>
> > Try adding sofia profile internal siptrace on to see the sip traffic too.
>
>
> Call log with sip trace http://pastebin.freeswitch.org/13065
>
>
> Thanks again :)
>
>
>
> --
>
> Regard's
>
> Sergey Scheglov
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <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
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From sid at eltc.ru  Thu May 27 11:49:04 2010
From: sid at eltc.ru (Sergey Scheglov)
Date: Fri, 28 May 2010 01:49:04 +0700
Subject: [Freeswitch-users] Don't work playback after bypass media mode.
In-Reply-To: <AANLkTimUWPk_SoDoIRHbkfISJz0ibNfyWLulZxsn9gQ0@mail.gmail.com>
References: <1427456515.20100526174916@eltc.ru>
	<C7E0DD6C-0EBB-4C1F-A47A-D3685DF5C6FC@jerris.com>
	<1751544061.20100527132304@eltc.ru>
	<AANLkTimUWPk_SoDoIRHbkfISJz0ibNfyWLulZxsn9gQ0@mail.gmail.com>
Message-ID: <1853205587.20100528014904@eltc.ru>

Hi, Anthony.

Thank's for reply.

You wrote 27 may 2010 ?., 20:34:15:

> playback executes on line 87 of your trace.

yes, but executed by log after 30 seconds after line 84 and immediately hangup.
Note, duration wav file - 4 sec.

> If you do not hear the audio, It means the re-establishment of media fails
> somehow based on your topology or
> the phone you are on does not support early media, change your pre-answer
> before the playback to answer to verify.

I don't hear the audio, because FS don't send RTP traffic to my phone in early media mode (checked sniffers) in my case.
If dialplans is: 
<extension name="fail_balance">
<condition field="${blabla}" expression="^\-1$">
<action application="pre_answer"/>
<action application="sleep" data="1000"/>
<action application="playback" data="elight/neg_balance.wav"/>
<action application="hangup" data="BEARERCAPABILITY_NOTAUTH"/>
</condition>
</extension>
then work's fine, no problems (bypass_media=true not set). 

If set answer, problem disappears. But answer it's not for my case.

> Try adding sofia profile internal siptrace on to see the sip traffic too.

Call log with sip trace http://pastebin.freeswitch.org/13065

Thanks again :)

Best regard's
Sergey Scheglov




From sid at eltc.ru  Thu May 27 12:12:28 2010
From: sid at eltc.ru (Sergey Scheglov)
Date: Fri, 28 May 2010 02:12:28 +0700
Subject: [Freeswitch-users] Don't work playback after bypass media mode.
In-Reply-To: <AANLkTinpkGvmvUvOQzAQdmUHHMNN_kxWQY304wlMEBwC@mail.gmail.com>
References: <1427456515.20100526174916@eltc.ru>
	<C7E0DD6C-0EBB-4C1F-A47A-D3685DF5C6FC@jerris.com>
	<1751544061.20100527132304@eltc.ru>
	<AANLkTimUWPk_SoDoIRHbkfISJz0ibNfyWLulZxsn9gQ0@mail.gmail.com>
	<141728530.20100528013547@eltc.ru>
	<AANLkTinpkGvmvUvOQzAQdmUHHMNN_kxWQY304wlMEBwC@mail.gmail.com>
Message-ID: <55492769.20100528021228@eltc.ru>

Hi, Anthony.

Thanks, thanks, thanks ))
You wrote 28 may 2010., 1:46:05:

> it's against the SIP spec to re-negotiate media before you have answered.
> the transaction to negotiate the SDP from the original call has not
> completed so it's illegal to send a re-invite.

Do I understand correctly that the dialplan is not correct?

> instead you should use bypass_media_after_bridge=true so the bypass only
> happens when the bridge works.

As I wrote in first message:
"If set bypass_media_after_bridge instead bypass_media, then works
fine, BUT changing codec negotiation." And two phones that support
codecs, which not present in internal profile, not be able to call
each other.

Best regards
Sergey Scheglov




From msc at freeswitch.org  Thu May 27 12:28:53 2010
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 May 2010 12:28:53 -0700
Subject: [Freeswitch-users] Get on IRC day!!
Message-ID: <AANLkTiloCcnJzCGqmVInVKAz4GayhcbHRojcQmaMwNQy@mail.gmail.com>

Okay everyone, please join #freeswitch on IRC! We want everyone to come hang
out with us. Also, we have many new ones coming in asking for advice. Please
join up and help spread the knowledge.

See you online!
-Michael (IRC: mercutioviz)
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From anthony.minessale at gmail.com  Thu May 27 13:50:37 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 May 2010 15:50:37 -0500
Subject: [Freeswitch-users] Don't work playback after bypass media mode.
In-Reply-To: <55492769.20100528021228@eltc.ru>
References: <1427456515.20100526174916@eltc.ru>
	<C7E0DD6C-0EBB-4C1F-A47A-D3685DF5C6FC@jerris.com>
	<1751544061.20100527132304@eltc.ru>
	<AANLkTimUWPk_SoDoIRHbkfISJz0ibNfyWLulZxsn9gQ0@mail.gmail.com>
	<141728530.20100528013547@eltc.ru>
	<AANLkTinpkGvmvUvOQzAQdmUHHMNN_kxWQY304wlMEBwC@mail.gmail.com>
	<55492769.20100528021228@eltc.ru>
Message-ID: <AANLkTim8UvXmpx0udG-krpkdp9zDrtt5Gk-InKp2AjBF@mail.gmail.com>

not sure what you mean but i'm simply explaining to you that you cannot
re-invite to change the media in the SOA before the initial call was setup.


On Thu, May 27, 2010 at 2:12 PM, Sergey Scheglov <sid at eltc.ru> wrote:

> Hi, Anthony.
>
> Thanks, thanks, thanks ))
> You wrote 28 may 2010., 1:46:05:
>
> > it's against the SIP spec to re-negotiate media before you have answered.
> > the transaction to negotiate the SDP from the original call has not
> > completed so it's illegal to send a re-invite.
>
> Do I understand correctly that the dialplan is not correct?
>
> > instead you should use bypass_media_after_bridge=true so the bypass only
> > happens when the bridge works.
>
> As I wrote in first message:
> "If set bypass_media_after_bridge instead bypass_media, then works
> fine, BUT changing codec negotiation." And two phones that support
> codecs, which not present in internal profile, not be able to call
> each other.
>
> Best regards
> Sergey Scheglov
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <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
-------------- next part --------------
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From phone.bytes at gmail.com  Thu May 27 14:51:38 2010
From: phone.bytes at gmail.com (Phone)
Date: Thu, 27 May 2010 15:51:38 -0600
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
In-Reply-To: <AANLkTikGQlZ4N3zfYv18COvbubEi-8LSOZxnAy1KC5sP@mail.gmail.com>
References: <490aedfdd7c8ac680bbe0d7f68c786de@mail.gmail.com>	<4BFBE6C2.5000703@gmail.com>	<AANLkTilsHn9znajzF0Jj-QDu4je9QW7CeymuBYE0SyKa@mail.gmail.com>	<4BFBEF7C.9020107@gmail.com>	<9250A9FF05DB4B879DEF6ECE8E3FFAC1@dell9400>	<AANLkTilU2OwgH0Qk_Qoc3R7ErPO_Z_VRhePusetkRkeW@mail.gmail.com>	<AANLkTikXWvEB_C3jqOWLHaVcns9-U5Hok-5MqgE_uzzZ@mail.gmail.com>	<4BFC4A64.5090208@gmail.com>	<AANLkTinFtVXa7CPou-ShPYIXHq0AktokhptGSEFKBkge@mail.gmail.com>	<D311BDCB06324D0DB77898FC6FF4B0EB@dell9400>	<AANLkTilAMe1lRKK4R6Nmr29FzfI-pP3wRa7Foabmd8Dp@mail.gmail.com>	<4BFD4562.5030804@gmail.com>
	<AANLkTikGQlZ4N3zfYv18COvbubEi-8LSOZxnAy1KC5sP@mail.gmail.com>
Message-ID: <4BFEE96A.2010003@gmail.com>

Thanks,

Examples and feedback are most helpful!

Bob Coleman wrote:
> Hi,
>
> Will incorporate some threading into the example I am working on for you.
>
> The outbound event socket method is very similar to the dialogic
> environment, I know how you felt though, but by starting small(like
> just even answering a call) gets you moving pretty quick. Freeswitch
> is lots of fun to work with, and the guys on here are very
> supportive!!
>
> Bob
>
> On Thu, May 27, 2010 at 3:59 AM, Phone <phone.bytes at gmail.com> wrote:
>   
>> Thanks to all for the most helpful feedback.  Sharing your approaches
>> and experiences are a big help.  I look forward to the upcoming code
>> samples.
>>
>> I was coming from a windows/dialogic environment where I used a library
>> that allowed me to work on a little higher level.  For example, I had a
>> call to "play a file" that took a parameter of whether or not to allow a
>> dtmf to interrupt.  There was also a call to "ReadDtmfs" that took
>> parameters to specify the number of Dtmf's to read, how long to wait for
>> them, and what terminating character to use.  I guess that you could
>> write some scripts or compiled code with these same types of functions
>> to simplify some of these routine tasks with reusable code?
>>
>> Also, the library handled the threading and scheduling with the OS.  I
>> am still unclear on handling the events.  I guess you have a big loop
>> reading events and then acting on them using the uuid to determine which
>> call it is and how to deal with the next step of the call?  Any feedback
>> on this part of the project?
>>
>> Again, Thanks!
>>
>> Bob Coleman wrote:
>>     
>>> Ah sorry, I started with the esl to get an understanding then wrote my
>>> own socket library(was actually very easy to do), when I mean docs I
>>> mean the event socket docs. I still think of it as the esl, my
>>> mistake.
>>>
>>> http://wiki.freeswitch.org/wiki/Event_Socket
>>>
>>> I started with a codeplex project, that had been abandoned, and then
>>> once I understood the structure of the event socket language, was able
>>> to rewrite it to better handle what we were doing.
>>>
>>> I also married it up to an old gotdotnet asterisk fast agi project,
>>> once again abandoned, to allow for the use of asterisk as well, but in
>>> the end freeswitch won because we could use just one platform.
>>>
>>> I am busy writing a small sample app at the moment to demonstrate a
>>> problem I am trying to solve. Can release that code once sorted. Will
>>> be in a week or so. Am intending it as a quick way of testing event
>>> sockets, and trying various commands etc. before commiting to coding
>>> something.
>>>
>>> Bob
>>>
>>> On Wed, May 26, 2010 at 2:26 PM, Jan Berger <jan.berger at video24.no> wrote:
>>>
>>>       
>>>> Hi,
>>>>
>>>> Do you have some sample code you could share + what docs did you look at?
>>>>
>>>> I would like to write and test some C# using ESL for my own work.
>>>>
>>>> Jan
>>>>
>>>> -----Original Message-----
>>>> From: freeswitch-users-bounces at lists.freeswitch.org
>>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bob
>>>> Coleman
>>>> Sent: 26. mai 2010 04:07
>>>> To: freeswitch-users at lists.freeswitch.org
>>>> Subject: Re: [Freeswitch-users] Questions on Building an application for
>>>> FreeSWITCH
>>>>
>>>> We used c# as the rest of our systems are windows based. The language
>>>> doesnt matter too much, as long as you know where you are headed, what
>>>> performance you require, and what platform you are going to be using.
>>>>
>>>> Found the ESL so much easier than the dialogic c library we were using.
>>>>
>>>> The docs for the esl are easy to understand, the thing I couldnt get
>>>> my head around initially was the dialing out, with the dialogic you
>>>> are in the middle when you dial, ie already on the channel, but with
>>>> freeswitch you are kind of the third party when you dial, the channel
>>>> being created by the dialing and handing it off to be worked on. We
>>>> make the call via an inbound event socket and hand it off to an
>>>> outbound event socket application via the dialplan.
>>>>
>>>> On Wed, May 26, 2010 at 10:08 AM, Phone <phone.bytes at gmail.com> wrote:
>>>>
>>>>         
>>>>> Thanks for the info.  What language did you use?
>>>>>
>>>>> Bob Coleman wrote:
>>>>>
>>>>>           
>>>>>> Have just recently completed a project to convert an old windows
>>>>>> dialogic application(vb6) to FreeSWITCH, would reccommend using the
>>>>>> ESL, was able to map the old dialogic calls to the ESL calls pretty
>>>>>> easily. We used a mixture of inbound and outbound sockets, as we have
>>>>>> people dialing us, not just dialing out etc.
>>>>>>
>>>>>> With the dialogic you open a port and make the call and handle the
>>>>>> dtmf, with freeswitch you create a socket connection to FreeSWITCH to
>>>>>> dial the number and then hand it off to an extension for processing
>>>>>> the dtmf(that is one approach any way)
>>>>>>
>>>>>> Bob
>>>>>>
>>>>>> On Wed, May 26, 2010 at 6:45 AM, Michael Collins <msc at freeswitch.org>
>>>>>>
>>>>>>             
>>>> wrote:
>>>>
>>>>         
>>>>>>> On Tue, May 25, 2010 at 11:38 AM, Jan Berger <jan.berger at video24.no>
>>>>>>>
>>>>>>>               
>>>> wrote:
>>>>
>>>>         
>>>>>>>> Actually - before you get "to smart" - may I suggest that you start
>>>>>>>> writing
>>>>>>>> - or improving - the getting started sections of the doc. Address the
>>>>>>>> areas
>>>>>>>> where you struggle and let others benefit from your work.
>>>>>>>>
>>>>>>>> I have been through similar issues myself - FS is one of the easier
>>>>>>>> projects
>>>>>>>> to work with once you get under the hood, but you basically need to
>>>>>>>>
>>>>>>>>                 
>>>> evolve
>>>>
>>>>         
>>>>>>>> to the level where you read the source code.
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>                 
>>>>>>> And if you can wait 2+ months for "the book" then that should help as
>>>>>>>
>>>>>>>               
>>>> well.
>>>>
>>>>         
>>>>>>> :D
>>>>>>> -MC
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>               
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>>             
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>           
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>         
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>       
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From codecomplete at free.fr  Thu May 27 16:51:46 2010
From: codecomplete at free.fr (GillesToo)
Date: Thu, 27 May 2010 16:51:46 -0700 (PDT)
Subject: [Freeswitch-users] Installing OpenZap?
Message-ID: <1275004306588-5111050.post@n2.nabble.com>


Hello

I read the documents in the wiki, but my Dahdi + OpenZap + FXO card combo
doesn't work, so I'd like to check if I made a mistake, overlooked
something, or maybe some things have changed since Asterisk moved from
Zaptel to Dahdi and the wiki hasn't been updated accordingly:

1. Compiled Dahdi 2.3.0, and successfully loaded it:

# /etc/init.d/dahdi status
### Span  1: WCFXO/0 "Wildcard X100P Board 1" (MASTER) RED
  1 FXO        FXSKS       (SWEC: MG2)  RED
  
2. vi /usr/local/freeswitch/conf/zt.conf: Nothing to do here, apparently

3. vi /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml:
Uncommented <load module ="mod_openzap"/> 

4. vi /usr/local/freeswitch/conf/autoload_configs/openzap.conf.xml: Left
default settings

5. vi /usr/local/freeswitch/conf/openzap.conf:
[span zt]
name => OpenZAP
number => 1
fxo-channel => 1

6. The wiki says to then run fs_cli, followed by "oz list", but: "oz list:
Command not found!"

Did I miss something?

Thank you for any hint.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Installing-OpenZap-tp5111050p5111050.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From delorenzodesign at gmail.com  Thu May 27 18:19:13 2010
From: delorenzodesign at gmail.com (Michael De Lorenzo)
Date: Thu, 27 May 2010 21:19:13 -0400
Subject: [Freeswitch-users] 407 Proxy Authentication
Message-ID: <AANLkTikJ9zW_yH0hH0GrVK5ovFbcbY9XOie-54h55f4P@mail.gmail.com>

I've tried updating our ACL conf, but that doesn't seem to help as I still
get the 407 error (line 50 from the Gist below).  Am I correct that the
authentication is failing when Verizon attempts to contact our switch? Or is
it an authentication failure when I hit Verizon?

Here's the trace of the call attempt (I've replaced an IP address and phone
number, so I do realize that they're not correct) -
http://gist.github.com/416604

send 1129 bytes to udp/[63.79.178.192]:5060 at 01:04:26.090037:
   ------------------------------------------------------------------------
   INVITE sip:17895551234 at 63.79.178.192
<sip%3A17895551234 at 63.79.178.192> SIP/2.0
   Via: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK0NZjmNS5Byj4c
   Max-Forwards: 70
   From: "C3 Mgmt" <sip:9727289377 at 63.79.178.192
<sip%3A9727289377 at 63.79.178.192>;transport=udp>;tag=tgHN0ScKyNcXr
   To: <sip:17895551234 at 63.79.178.192 <sip%3A17895551234 at 63.79.178.192>>
   Call-ID: cd26b853-e497-122d-278a-000bdb94aab9
   CSeq: 131385997 INVITE
   Contact: <sip:gw+verizon at 0.0.0.0:5080;transport=udp;gw=verizon>
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-28484a1 2010-04-22
15:06:05 -0400
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, hold, refer
   Privacy: none
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 291
   X-FS-Support: update_display
   P-Asserted-Identity: "C3 Mgmt" <sip:+19727289377 at 63.79.178.192
<sip%3A%2B19727289377 at 63.79.178.192>>

   v=0
   o=FreeSWITCH 1274985280 1274985281 IN IP4 0.0.0.0
   s=FreeSWITCH
   c=IN IP4 0.0.0.0
   t=0 0
   m=audio 21600 RTP/AVP 0 8 3 101 13
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:3 GSM/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=rtpmap:13 CN/8000
   a=ptime:20
   ------------------------------------------------------------------------
2010-05-27 21:00:10.128406 [DEBUG] sofia.c:4168 Channel
sofia/external/17895551234 entering state [calling][0]
recv 296 bytes from udp/[63.79.178.192]:5060 at 01:00:10.192551:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   v: SIP/2.0/UDP
0.0.0.0:5080;rport;branch=z9hG4bKZc6Sjt81eNvHH;received=0.0.0.0
   f: "C3 Mgmt" <sip:9727289377 at 63.79.178.192
<sip%3A9727289377 at 63.79.178.192>;transport=udp>;tag=S7QvyyUF1cpaD
   t: <sip:17895551234 at 63.79.178.192 <sip%3A17895551234 at 63.79.178.192>>
   i: 348b8efb-e497-122d-278a-000bdb94aab9
   CSeq: 131385870 INVITE
   l: 0

   ------------------------------------------------------------------------
recv 425 bytes from udp/[63.79.178.192]:5060 at 01:00:10.192710:
   ------------------------------------------------------------------------
   SIP/2.0 407 Proxy Authentication Required
   v: SIP/2.0/UDP
0.0.0.0:5080;rport;branch=z9hG4bKZc6Sjt81eNvHH;received=0.0.0.0
   f: "C3 Mgmt" <sip:9727289377 at 63.79.178.192
<sip%3A9727289377 at 63.79.178.192>;transport=udp>;tag=S7QvyyUF1cpaD
   t: <sip:17895551234 at 63.79.178.192
<sip%3A17895551234 at 63.79.178.192>>;tag=22703bab
   i: 348b8efb-e497-122d-278a-000bdb94aab9
   CSeq: 131385870 INVITE
   l: 0
   Proxy-Authenticate: DIGEST
realm="WCOM",nonce="a586274e395fb9b6f66f1a7829b4531a.1275008409"

   ------------------------------------------------------------------------
send 350 bytes to udp/[63.79.178.192]:5060 at 01:00:10.192977:
   ------------------------------------------------------------------------
   ACK sip:17895551234 at 63.79.178.192 <sip%3A17895551234 at 63.79.178.192> SIP/2.0
   Via: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bKZc6Sjt81eNvHH
   Max-Forwards: 70
   From: "C3 Mgmt" <sip:9727289377 at 63.79.178.192
<sip%3A9727289377 at 63.79.178.192>;transport=udp>;tag=S7QvyyUF1cpaD
   To: <sip:17895551234 at 63.79.178.192
<sip%3A17895551234 at 63.79.178.192>>;tag=22703bab
   Call-ID: 348b8efb-e497-122d-278a-000bdb94aab9
   CSeq: 131385870 ACK
   Content-Length: 0

   ------------------------------------------------------------------------
2010-05-27 21:00:10.193015 [DEBUG] sofia.c:4168 Channel
sofia/external/17895551234 entering state [terminated][904]
2010-05-27 21:00:10.193015 [NOTICE] sofia.c:4804 Hangup
sofia/external/17895551234 [CS_CONSUME_MEDIA] [NORMAL_UNSPECIFIED]
2010-05-27 21:00:10.193015 [DEBUG] switch_channel.c:2117 Send signal
sofia/external/17895551234 [KILL]
2010-05-27 21:00:10.193015 [DEBUG] switch_core_session.c:1022 Send
signal sofia/external/17895551234 [BREAK]
2010-05-27 21:00:10.193015 [DEBUG] switch_core_state_machine.c:314
(sofia/external/17895551234) Running State Change CS_HANGUP
2010-05-27 21:00:10.193015 [DEBUG] switch_core_state_machine.c:499
(sofia/external/17895551234) State HANGUP
2010-05-27 21:00:10.193015 [DEBUG] mod_sofia.c:410
sofia/external/17895551234 Overriding SIP cause 480 with 904 from the
other leg
2010-05-27 21:00:10.193015 [DEBUG] mod_sofia.c:416 Channel
sofia/external/17895551234 hanging up, cause: NORMAL_UNSPECIFIED
2010-05-27 21:00:10.193015 [DEBUG] switch_core_state_machine.c:46
sofia/external/17895551234 Standard HANGUP, cause: NORMAL_UNSPECIFIED
2010-05-27 21:00:10.193015 [DEBUG] switch_core_state_machine.c:499
(sofia/external/17895551234) State HANGUP going to sleep
2010-05-27 21:00:10.193015 [DEBUG] switch_core_state_machine.c:333
(sofia/external/17895551234) State Change CS_HANGUP -> CS_REPORTING
2010-05-27 21:00:10.193015 [DEBUG] switch_core_session.c:1022 Send
signal sofia/external/17895551234 [BREAK]
2010-05-27 21:00:10.193015 [DEBUG] switch_core_state_machine.c:314
(sofia/external/17895551234) Running State Change CS_REPORTING
2010-05-27 21:00:10.193015 [DEBUG] switch_ivr_originate.c:3228
Originate Resulted in Error Cause: 31 [NORMAL_UNSPECIFIED]
2010-05-27 21:00:10.193015 [DEBUG] switch_core_state_machine.c:590
(sofia/external/17895551234) State REPORTING


freeswitch at govinteract-fs-dev-2> 2010-05-27 21:00:10.299296 [DEBUG]
switch_core_state_machine.c:53 sofia/external/17895551234 Standard
REPORTING, cause: NORMAL_UNSPECIFIED
2010-05-27 21:00:10.299296 [DEBUG] switch_core_state_machine.c:590
(sofia/external/17895551234) State REPORTING going to sleep
2010-05-27 21:00:10.299296 [DEBUG] switch_core_state_machine.c:327
(sofia/external/17895551234) State Change CS_REPORTING -> CS_DESTROY
2010-05-27 21:00:10.299296 [DEBUG] switch_core_session.c:1022 Send
signal sofia/external/17895551234 [BREAK]
2010-05-27 21:00:10.299296 [DEBUG] switch_core_session.c:1165 Session
6 (sofia/external/17895551234) Locked, Waiting on external entities
2010-05-27 21:00:10.299296 [NOTICE] switch_core_session.c:1183 Session
6 (sofia/external/17895551234) Ended
2010-05-27 21:00:10.299296 [NOTICE] switch_core_session.c:1185 Close
Channel sofia/external/17895551234 [CS_DESTROY]
2010-05-27 21:00:10.299296 [DEBUG] switch_core_state_machine.c:428
(sofia/external/17895551234) Running State Change CS_DESTROY
2010-05-27 21:00:10.299296 [DEBUG] switch_core_state_machine.c:439
(sofia/external/17895551234) State DESTROY
2010-05-27 21:00:10.299296 [DEBUG] mod_sofia.c:343
sofia/external/17895551234 SOFIA DESTROY
2010-05-27 21:00:10.299296 [DEBUG] switch_core_state_machine.c:60
sofia/external/17895551234 Standard DESTROY
2010-05-27 21:00:10.299296 [DEBUG] switch_core_state_machine.c:439
(sofia/external/17895551234) State DESTROY going to sleep

freeswitch at govinteract-fs-dev-2> lua test1.lua
2010-05-27 21:04:26.087322 [DEBUG] switch_ivr_originate.c:1885
variable string 0 = [sip_cid_type=pid]
2010-05-27 21:04:26.087322 [DEBUG] switch_ivr_originate.c:1885
variable string 1 = [origination_caller_id_name=C3 Mgmt]
2010-05-27 21:04:26.087322 [DEBUG] switch_ivr_originate.c:1885
variable string 2 = [origination_caller_id_number=+19727289377]
2010-05-27 21:04:26.087322 [DEBUG] switch_ivr_originate.c:1885
variable string 3 = [ignore_early_media=true]
2010-05-27 21:04:26.088360 [NOTICE] switch_channel.c:669 New Channel
sofia/external/17895551234 [f5c8454e-69f4-11df-b69d-099cfc924087]
2010-05-27 21:04:26.088360 [DEBUG] mod_sofia.c:3444
(sofia/external/17895551234) State Change CS_NEW -> CS_INIT
2010-05-27 21:04:26.088360 [DEBUG] switch_core_session.c:1022 Send
signal sofia/external/17895551234 [BREAK]
2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:314
(sofia/external/17895551234) Running State Change CS_INIT
2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:338
(sofia/external/17895551234) State INIT
2010-05-27 21:04:26.088360 [DEBUG] mod_sofia.c:83
sofia/external/17895551234 SOFIA INIT
2010-05-27 21:04:26.088360 [DEBUG] mod_sofia.c:117
(sofia/external/17895551234) State Change CS_INIT -> CS_ROUTING
2010-05-27 21:04:26.088360 [DEBUG] switch_core_session.c:1022 Send
signal sofia/external/17895551234 [BREAK]
2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:338
(sofia/external/17895551234) State INIT going to sleep
2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:314
(sofia/external/17895551234) Running State Change CS_ROUTING
2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:341
(sofia/external/17895551234) State ROUTING
2010-05-27 21:04:26.088360 [DEBUG] mod_sofia.c:140
sofia/external/17895551234 SOFIA ROUTING
2010-05-27 21:04:26.088360 [DEBUG] switch_ivr_originate.c:66
(sofia/external/17895551234) State Change CS_ROUTING ->
CS_CONSUME_MEDIA
2010-05-27 21:04:26.088360 [DEBUG] switch_core_session.c:1022 Send
signal sofia/external/17895551234 [BREAK]
2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:341
(sofia/external/17895551234) State ROUTING going to sleep
2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:314
(sofia/external/17895551234) Running State Change CS_CONSUME_MEDIA
2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:360
(sofia/external/17895551234) State CONSUME_MEDIA
2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:360
(sofia/external/17895551234) State CONSUME_MEDIA going to sleep
send 1129 bytes to udp/[63.79.178.192]:5060 at 01:04:26.090037:
   ------------------------------------------------------------------------
   INVITE sip:17895551234 at 63.79.178.192
<sip%3A17895551234 at 63.79.178.192> SIP/2.0
   Via: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK0NZjmNS5Byj4c
   Max-Forwards: 70
   From: "C3 Mgmt" <sip:9727289377 at 63.79.178.192
<sip%3A9727289377 at 63.79.178.192>;transport=udp>;tag=tgHN0ScKyNcXr
   To: <sip:17895551234 at 63.79.178.192 <sip%3A17895551234 at 63.79.178.192>>
   Call-ID: cd26b853-e497-122d-278a-000bdb94aab9
   CSeq: 131385997 INVITE
   Contact: <sip:gw+verizon at 0.0.0.0:5080;transport=udp;gw=verizon>
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-28484a1 2010-04-22
15:06:05 -0400
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, hold, refer
   Privacy: none
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 291
   X-FS-Support: update_display
   P-Asserted-Identity: "C3 Mgmt" <sip:+19727289377 at 63.79.178.192
<sip%3A%2B19727289377 at 63.79.178.192>>

   v=0
   o=FreeSWITCH 1274985280 1274985281 IN IP4 0.0.0.0
   s=FreeSWITCH
   c=IN IP4 0.0.0.0
   t=0 0
   m=audio 23386 RTP/AVP 0 8 3 101 13
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:3 GSM/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=rtpmap:13 CN/8000
   a=ptime:20
   ------------------------------------------------------------------------
2010-05-27 21:04:26.089756 [DEBUG] sofia.c:4168 Channel
sofia/external/17895551234 entering state [calling][0]
recv 425 bytes from udp/[63.79.178.192]:5060 at 01:04:26.167603:
   ------------------------------------------------------------------------
   SIP/2.0 407 Proxy Authentication Required
   v: SIP/2.0/UDP
0.0.0.0:5080;rport;branch=z9hG4bK0NZjmNS5Byj4c;received=0.0.0.0
   f: "C3 Mgmt" <sip:9727289377 at 63.79.178.192
<sip%3A9727289377 at 63.79.178.192>;transport=udp>;tag=tgHN0ScKyNcXr
   t: <sip:17895551234 at 63.79.178.192
<sip%3A17895551234 at 63.79.178.192>>;tag=41e44fd5
   i: cd26b853-e497-122d-278a-000bdb94aab9
   CSeq: 131385997 INVITE
   l: 0
   Proxy-Authenticate: DIGEST
realm="WCOM",nonce="43ea850af91d419e72d85c0375a60237.1275008665"

   ------------------------------------------------------------------------
send 350 bytes to udp/[63.79.178.192]:5060 at 01:04:26.167913:
   ------------------------------------------------------------------------
   ACK sip:17895551234 at 63.79.178.192 <sip%3A17895551234 at 63.79.178.192> SIP/2.0
   Via: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK0NZjmNS5Byj4c
   Max-Forwards: 70
   From: "C3 Mgmt" <sip:9727289377 at 63.79.178.192
<sip%3A9727289377 at 63.79.178.192>;transport=udp>;tag=tgHN0ScKyNcXr
   To: <sip:17895551234 at 63.79.178.192
<sip%3A17895551234 at 63.79.178.192>>;tag=41e44fd5
   Call-ID: cd26b853-e497-122d-278a-000bdb94aab9
   CSeq: 131385997 ACK
   Content-Length: 0

   ------------------------------------------------------------------------
send 1359 bytes to udp/[63.79.178.192]:5060 at 01:04:26.168477:
   ------------------------------------------------------------------------
   INVITE sip:17895551234 at 63.79.178.192
<sip%3A17895551234 at 63.79.178.192> SIP/2.0
   Via: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK1yrBpga9868pr
   Max-Forwards: 70
   From: "C3 Mgmt" <sip:9727289377 at 63.79.178.192
<sip%3A9727289377 at 63.79.178.192>;transport=udp>;tag=tgHN0ScKyNcXr
   To: <sip:17895551234 at 63.79.178.192 <sip%3A17895551234 at 63.79.178.192>>
   Call-ID: cd26b853-e497-122d-278a-000bdb94aab9
   CSeq: 131385998 INVITE
   Contact: <sip:gw+verizon at 0.0.0.0:5080;transport=udp;gw=verizon>
   Expires: 3600
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-28484a1 2010-04-22
15:06:05 -0400
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, hold, refer
   Proxy-Authorization: Digest username="9727289377", realm="WCOM",
nonce="43ea850af91d419e72d85c0375a60237.1275008665", algorithm=MD5,
uri="sip:17895551234 at 63.79.178.192 <sip%3A17895551234 at 63.79.178.192>",
response="1871a9dbd248f92b65df2af84c6057f8"
   Privacy: none
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 291
   X-FS-Support: update_display
   P-Asserted-Identity: "C3 Mgmt" <sip:+19727289377 at 63.79.178.192
<sip%3A%2B19727289377 at 63.79.178.192>>

   v=0
   o=FreeSWITCH 1274985280 1274985281 IN IP4 0.0.0.0
   s=FreeSWITCH
   c=IN IP4 0.0.0.0
   t=0 0
   m=audio 23386 RTP/AVP 0 8 3 101 13
   a=rtpmap:0 PCMU/8000
   a=rtpmap:8 PCMA/8000
   a=rtpmap:3 GSM/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=rtpmap:13 CN/8000
   a=ptime:20
   ------------------------------------------------------------------------
2010-05-27 21:04:26.168049 [DEBUG] sofia.c:4168 Channel
sofia/external/17895551234 entering state [calling][0]
recv 296 bytes from udp/[63.79.178.192]:5060 at 01:04:26.231231:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   v: SIP/2.0/UDP
0.0.0.0:5080;rport;branch=z9hG4bK1yrBpga9868pr;received=0.0.0.0
   f: "C3 Mgmt" <sip:9727289377 at 63.79.178.192
<sip%3A9727289377 at 63.79.178.192>;transport=udp>;tag=tgHN0ScKyNcXr
   t: <sip:17895551234 at 63.79.178.192 <sip%3A17895551234 at 63.79.178.192>>
   i: cd26b853-e497-122d-278a-000bdb94aab9
   CSeq: 131385998 INVITE
   l: 0

   ------------------------------------------------------------------------
recv 425 bytes from udp/[63.79.178.192]:5060 at 01:04:26.231408:
   ------------------------------------------------------------------------
   SIP/2.0 407 Proxy Authentication Required
   v: SIP/2.0/UDP
0.0.0.0:5080;rport;branch=z9hG4bK1yrBpga9868pr;received=0.0.0.0
   f: "C3 Mgmt" <sip:9727289377 at 63.79.178.192
<sip%3A9727289377 at 63.79.178.192>;transport=udp>;tag=tgHN0ScKyNcXr
   t: <sip:17895551234 at 63.79.178.192
<sip%3A17895551234 at 63.79.178.192>>;tag=1da20439
   i: cd26b853-e497-122d-278a-000bdb94aab9
   CSeq: 131385998 INVITE
   l: 0
   Proxy-Authenticate: DIGEST
realm="WCOM",nonce="43ea850af91d419e72d85c0375a60237.1275008665"

   ------------------------------------------------------------------------
send 350 bytes to udp/[63.79.178.192]:5060 at 01:04:26.231667:
   ------------------------------------------------------------------------
   ACK sip:17895551234 at 63.79.178.192 <sip%3A17895551234 at 63.79.178.192> SIP/2.0
   Via: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK1yrBpga9868pr
   Max-Forwards: 70
   From: "C3 Mgmt" <sip:9727289377 at 63.79.178.192
<sip%3A9727289377 at 63.79.178.192>;transport=udp>;tag=tgHN0ScKyNcXr
   To: <sip:17895551234 at 63.79.178.192
<sip%3A17895551234 at 63.79.178.192>>;tag=1da20439
   Call-ID: cd26b853-e497-122d-278a-000bdb94aab9
   CSeq: 131385998 ACK
   Content-Length: 0

   ------------------------------------------------------------------------
2010-05-27 21:04:26.231294 [DEBUG] sofia.c:4168 Channel
sofia/external/17895551234 entering state [terminated][904]
2010-05-27 21:04:26.231294 [NOTICE] sofia.c:4804 Hangup
sofia/external/17895551234 [CS_CONSUME_MEDIA] [NORMAL_UNSPECIFIED]
2010-05-27 21:04:26.231294 [DEBUG] switch_channel.c:2117 Send signal
sofia/external/17895551234 [KILL]
2010-05-27 21:04:26.231294 [DEBUG] switch_core_session.c:1022 Send
signal sofia/external/17895551234 [BREAK]
2010-05-27 21:04:26.232334 [DEBUG] switch_core_state_machine.c:314
(sofia/external/17895551234) Running State Change CS_HANGUP
2010-05-27 21:04:26.232334 [DEBUG] switch_core_state_machine.c:499
(sofia/external/17895551234) State HANGUP
2010-05-27 21:04:26.232334 [DEBUG] mod_sofia.c:410
sofia/external/17895551234 Overriding SIP cause 480 with 904 from the
other leg
2010-05-27 21:04:26.232334 [DEBUG] mod_sofia.c:416 Channel
sofia/external/17895551234 hanging up, cause: NORMAL_UNSPECIFIED
2010-05-27 21:04:26.232334 [DEBUG] switch_core_state_machine.c:46
sofia/external/17895551234 Standard HANGUP, cause: NORMAL_UNSPECIFIED
2010-05-27 21:04:26.232334 [DEBUG] switch_core_state_machine.c:499
(sofia/external/17895551234) State HANGUP going to sleep
2010-05-27 21:04:26.232334 [DEBUG] switch_core_state_machine.c:333
(sofia/external/17895551234) State Change CS_HANGUP -> CS_REPORTING
2010-05-27 21:04:26.232334 [DEBUG] switch_core_session.c:1022 Send
signal sofia/external/17895551234 [BREAK]
2010-05-27 21:04:26.232334 [DEBUG] switch_core_state_machine.c:314
(sofia/external/17895551234) Running State Change CS_REPORTING
2010-05-27 21:04:26.232334 [DEBUG] switch_ivr_originate.c:3228
Originate Resulted in Error Cause: 31 [NORMAL_UNSPECIFIED]
2010-05-27 21:04:26.232334 [DEBUG] switch_core_state_machine.c:590
(sofia/external/17895551234) State REPORTING


freeswitch at govinteract-fs-dev-2> 2010-05-27 21:04:26.260365 [DEBUG]
switch_core_state_machine.c:53 sofia/external/17895551234 Standard
REPORTING, cause: NORMAL_UNSPECIFIED
2010-05-27 21:04:26.260365 [DEBUG] switch_core_state_machine.c:590
(sofia/external/17895551234) State REPORTING going to sleep
2010-05-27 21:04:26.260365 [DEBUG] switch_core_state_machine.c:327
(sofia/external/17895551234) State Change CS_REPORTING -> CS_DESTROY
2010-05-27 21:04:26.260365 [DEBUG] switch_core_session.c:1022 Send
signal sofia/external/17895551234 [BREAK]
2010-05-27 21:04:26.260365 [DEBUG] switch_core_session.c:1165 Session
7 (sofia/external/17895551234) Locked, Waiting on external entities
2010-05-27 21:04:26.260365 [NOTICE] switch_core_session.c:1183 Session
7 (sofia/external/17895551234) Ended
2010-05-27 21:04:26.260365 [NOTICE] switch_core_session.c:1185 Close
Channel sofia/external/17895551234 [CS_DESTROY]
2010-05-27 21:04:26.260365 [DEBUG] switch_core_state_machine.c:428
(sofia/external/17895551234) Running State Change CS_DESTROY
2010-05-27 21:04:26.260365 [DEBUG] switch_core_state_machine.c:439
(sofia/external/17895551234) State DESTROY
2010-05-27 21:04:26.260365 [DEBUG] mod_sofia.c:343
sofia/external/17895551234 SOFIA DESTROY
2010-05-27 21:04:26.260365 [DEBUG] switch_core_state_machine.c:60
sofia/external/17895551234 Standard DESTROY
2010-05-27 21:04:26.260365 [DEBUG] switch_core_state_machine.c:439
(sofia/external/17895551234) State DESTROY going to sleep

Here's how I've edited our acl.conf.xml file:

<configuration name="acl.conf" description="Network Lists">
  <network-lists>
    <!--
         These ACL's are automatically created on startup.

         rfc1918.auto  - RFC1918 Space
         nat.auto      - RFC1918 Excluding your local lan.
         localnet.auto - ACL for your local lan.
         loopback.auto - ACL for your local lan.
    -->

    <list name="lan" default="allow">
      <node type="deny" cidr="192.168.42.0/24"/>
      <node type="allow" cidr="192.168.42.42/32"/>
   </list>
   <list name="verizon" default="deny">
      <node type="allow" cidr="63.79.178.192"/>
   </list>
   .... truncated, the rest is the default file ....

<http://gist.github.com/416604>Any help would be greatly appreciated!


2010/5/27 <freeswitch-users-request at lists.freeswitch.org>

> Send FreeSWITCH-users mailing list submissions to
>        freeswitch-users at lists.freeswitch.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
>        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> or, via email, send a message with subject or body 'help' to
>        freeswitch-users-request at lists.freeswitch.org
>
> You can reach the person managing the list at
>        freeswitch-users-owner at lists.freeswitch.org
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of FreeSWITCH-users digest..."
>
> Today's Topics:
>
>   1. Re: Don't work playback after bypass media mode. (Sergey Scheglov)
>   2. Re: Don't work playback after bypass media mode. (Sergey Scheglov)
>   3. Re: Don't work playback after bypass media mode.
>      (Anthony Minessale)
>   4. Re: Don't work playback after bypass media mode. (Sergey Scheglov)
>   5. Re: Don't work playback after bypass media mode. (Sergey Scheglov)
>
>
> ---------- Forwarded message ----------
> From: Sergey Scheglov <sid at eltc.ru>
> To: freeswitch-users at lists.freeswitch.org
> Date: Fri, 28 May 2010 01:31:26 +0700
> Subject: Re: [Freeswitch-users] Don't work playback after bypass media
> mode.
>
> Hi, Anthony.
>
>
> Thank's for reply.
>
>
> You wrote 27 may 2010 ?., 20:34:15:
>
>
> > playback executes on line 87 of your trace.
>
>
> yes, but executed by log after 30 seconds after line 84 and immediately
> hangup.
>
> Note, duration wav file - 4 sec.
>
>
> > If you do not hear the audio, It means the re-establishment of media
> fails somehow based on your topology or
>
> the phone you are on does not support early media,
>
>
> I don't  hear the audio, because FS don't send RTP traffic to my phone in
> early media mode (checked sniffers) in my case.
>
> If dialplans is:
>
> <extension name="fail_balance">
>
>       <condition field="${blabla}" expression="^\-1$">
>
>         <action application="pre_answer"/>
>
>         <action application="sleep" data="1000"/>
>
>         <action application="playback" data="elight/neg_balance.wav"/>
>
>         <action application="hangup" data="BEARERCAPABILITY_NOTAUTH"/>
>
>       </condition>
>
>  </extension>
>
> then work's fine, no problems (bypass_media=true not set).
>
>
> > change your pre-answer before the playback to answer to verify.
>
>
> If set answer, problem disappears. But answer it's not for my case.
>
>
> > Try adding sofia profile internal siptrace on to see the sip traffic too.
>
>
> Call log with sip trace http://pastebin.freeswitch.org/13065
>
>
> Thanks again :)
>
>
> --
>
> Regard's
>
> Sergey Scheglov
>
>
> ---------- Forwarded message ----------
> From: Sergey Scheglov <sid at eltc.ru>
> To: Anthony Minessale <freeswitch-users at lists.freeswitch.org>
> Date: Fri, 28 May 2010 01:35:47 +0700
> Subject: Re: [Freeswitch-users] Don't work playback after bypass media
> mode.
>
> Hi, Anthony.
>
>
> You wrote 27 may 2010 ?., 20:34:15:
>
>
>
>  playback executes on line 87 of your trace.
>
>
> If you do not hear the audio, It means the re-establishment of media fails
> somehow based on your topology or
>
> the phone you are on does not support early media, change your pre-answer
> before the playback to answer to verify.
>
>
> Try adding sofia profile internal siptrace on to see the sip traffic too.
>
>
>
>
> > playback executes on line 87 of your trace.
>
>
> yes, but executed by log after 30 seconds after line 84 and immediately
> hangup.
>
> Note, duration wav file - 4 sec.
>
>
> > If you do not hear the audio, It means the re-establishment of media
> fails somehow based on your topology or
>
> the phone you are on does not support early media,
>
>
> I don't hear the audio, because FS don't send RTP traffic to my phone in
> early media mode (checked sniffers) in my case.
>
> If dialplans is:
>
> <extension name="fail_balance">
>
> <condition field="${blabla}" expression="^\-1$">
>
> <action application="pre_answer"/>
>
> <action application="sleep" data="1000"/>
>
> <action application="playback" data="elight/neg_balance.wav"/>
>
> <action application="hangup" data="BEARERCAPABILITY_NOTAUTH"/>
>
> </condition>
>
> </extension>
>
> then work's fine, no problems (bypass_media=true not set).
>
>
> > change your pre-answer before the playback to answer to verify.
>
>
> If set answer, problem disappears. But answer it's not for my case.
>
>
> > Try adding sofia profile internal siptrace on to see the sip traffic too.
>
>
> Call log with sip trace http://pastebin.freeswitch.org/13065
>
>
> Thanks again :)
>
>
>
> --
>
> Regard's
>
> Sergey Scheglov
>
>
> ---------- Forwarded message ----------
> From: Anthony Minessale <anthony.minessale at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Thu, 27 May 2010 13:46:05 -0500
> Subject: Re: [Freeswitch-users] Don't work playback after bypass media
> mode.
> it's against the SIP spec to re-negotiate media before you have answered.
> the transaction to negotiate the SDP from the original call has not
> completed so it's illegal to send a re-invite.
>
> instead you should use bypass_media_after_bridge=true so the bypass only
> happens when the bridge works.
>
>
> 2010/5/27 Sergey Scheglov <sid at eltc.ru>
>
>>  Hi, Anthony.
>>
>>
>> You wrote 27 may 2010 ?., 20:34:15:
>>
>>
>>
>>  playback executes on line 87 of your trace.
>>
>>
>> If you do not hear the audio, It means the re-establishment of media fails
>> somehow based on your topology or
>>
>> the phone you are on does not support early media, change your pre-answer
>> before the playback to answer to verify.
>>
>>
>> Try adding sofia profile internal siptrace on to see the sip traffic too.
>>
>>
>>
>>
>> > playback executes on line 87 of your trace.
>>
>>
>> yes, but executed by log after 30 seconds after line 84 and immediately
>> hangup.
>>
>> Note, duration wav file - 4 sec.
>>
>>
>> > If you do not hear the audio, It means the re-establishment of media
>> fails somehow based on your topology or
>>
>> the phone you are on does not support early media,
>>
>>
>> I don't hear the audio, because FS don't send RTP traffic to my phone in
>> early media mode (checked sniffers) in my case.
>>
>> If dialplans is:
>>
>> <extension name="fail_balance">
>>
>> <condition field="${blabla}" expression="^\-1$">
>>
>> <action application="pre_answer"/>
>>
>> <action application="sleep" data="1000"/>
>>
>> <action application="playback" data="elight/neg_balance.wav"/>
>>
>> <action application="hangup" data="BEARERCAPABILITY_NOTAUTH"/>
>>
>> </condition>
>>
>> </extension>
>>
>> then work's fine, no problems (bypass_media=true not set).
>>
>>
>> > change your pre-answer before the playback to answer to verify.
>>
>>
>> If set answer, problem disappears. But answer it's not for my case.
>>
>>
>> > Try adding sofia profile internal siptrace on to see the sip traffic
>> too.
>>
>>
>> Call log with sip trace http://pastebin.freeswitch.org/13065
>>
>>
>> Thanks again :)
>>
>>
>>
>> --
>>
>> Regard's
>>
>> Sergey Scheglov
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <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
>
>
> ---------- Forwarded message ----------
> From: Sergey Scheglov <sid at eltc.ru>
> To: Anthony Minessale <freeswitch-users at lists.freeswitch.org>
> Date: Fri, 28 May 2010 01:49:04 +0700
> Subject: Re: [Freeswitch-users] Don't work playback after bypass media
> mode.
> Hi, Anthony.
>
> Thank's for reply.
>
> You wrote 27 may 2010 ?., 20:34:15:
>
> > playback executes on line 87 of your trace.
>
> yes, but executed by log after 30 seconds after line 84 and immediately
> hangup.
> Note, duration wav file - 4 sec.
>
> > If you do not hear the audio, It means the re-establishment of media
> fails
> > somehow based on your topology or
> > the phone you are on does not support early media, change your pre-answer
> > before the playback to answer to verify.
>
> I don't hear the audio, because FS don't send RTP traffic to my phone in
> early media mode (checked sniffers) in my case.
> If dialplans is:
> <extension name="fail_balance">
> <condition field="${blabla}" expression="^\-1$">
> <action application="pre_answer"/>
> <action application="sleep" data="1000"/>
> <action application="playback" data="elight/neg_balance.wav"/>
> <action application="hangup" data="BEARERCAPABILITY_NOTAUTH"/>
> </condition>
> </extension>
> then work's fine, no problems (bypass_media=true not set).
>
> If set answer, problem disappears. But answer it's not for my case.
>
> > Try adding sofia profile internal siptrace on to see the sip traffic too.
>
> Call log with sip trace http://pastebin.freeswitch.org/13065
>
> Thanks again :)
>
> Best regard's
> Sergey Scheglov
>
>
>
>
>
>
> ---------- Forwarded message ----------
> From: Sergey Scheglov <sid at eltc.ru>
> To: Anthony Minessale <freeswitch-users at lists.freeswitch.org>
> Date: Fri, 28 May 2010 02:12:28 +0700
> Subject: Re: [Freeswitch-users] Don't work playback after bypass media
> mode.
> Hi, Anthony.
>
> Thanks, thanks, thanks ))
> You wrote 28 may 2010., 1:46:05:
>
> > it's against the SIP spec to re-negotiate media before you have answered.
> > the transaction to negotiate the SDP from the original call has not
> > completed so it's illegal to send a re-invite.
>
> Do I understand correctly that the dialplan is not correct?
>
> > instead you should use bypass_media_after_bridge=true so the bypass only
> > happens when the bridge works.
>
> As I wrote in first message:
> "If set bypass_media_after_bridge instead bypass_media, then works
> fine, BUT changing codec negotiation." And two phones that support
> codecs, which not present in internal profile, not be able to call
> each other.
>
> Best regards
> Sergey Scheglov
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From troy at tlainvestments.com  Thu May 27 19:17:54 2010
From: troy at tlainvestments.com (Troy Anderson)
Date: Thu, 27 May 2010 19:17:54 -0700
Subject: [Freeswitch-users] Installing OpenZap?
In-Reply-To: <1275004306588-5111050.post@n2.nabble.com>
References: <1275004306588-5111050.post@n2.nabble.com>
Message-ID: <6F3328D2-9169-4EE7-9CC5-7D649D7E4431@tlainvestments.com>

oz list will only work if openzap is loaded, so that's a clue that it isn't for some reason.

Try issuing reload mod_openzap from the cli and take a look at the output.  There is likely an error on load, and that error may lead you to what's wrong with the config.

-Troy

 
Troy Anderson
e troy at chronostelecom.com


On May 27, 2010, at 4:51 PM, GillesToo wrote:

> 
> Hello
> 
> I read the documents in the wiki, but my Dahdi + OpenZap + FXO card combo
> doesn't work, so I'd like to check if I made a mistake, overlooked
> something, or maybe some things have changed since Asterisk moved from
> Zaptel to Dahdi and the wiki hasn't been updated accordingly:
> 
> 1. Compiled Dahdi 2.3.0, and successfully loaded it:
> 
> # /etc/init.d/dahdi status
> ### Span  1: WCFXO/0 "Wildcard X100P Board 1" (MASTER) RED
>  1 FXO        FXSKS       (SWEC: MG2)  RED
> 
> 2. vi /usr/local/freeswitch/conf/zt.conf: Nothing to do here, apparently
> 
> 3. vi /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml:
> Uncommented <load module ="mod_openzap"/> 
> 
> 4. vi /usr/local/freeswitch/conf/autoload_configs/openzap.conf.xml: Left
> default settings
> 
> 5. vi /usr/local/freeswitch/conf/openzap.conf:
> [span zt]
> name => OpenZAP
> number => 1
> fxo-channel => 1
> 
> 6. The wiki says to then run fs_cli, followed by "oz list", but: "oz list:
> Command not found!"
> 
> Did I miss something?
> 
> Thank you for any hint.
> -- 
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Installing-OpenZap-tp5111050p5111050.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From sid at eltc.ru  Thu May 27 19:34:16 2010
From: sid at eltc.ru (Sergey Scheglov)
Date: Fri, 28 May 2010 09:34:16 +0700
Subject: [Freeswitch-users] Don't work playback after bypass media mode.
In-Reply-To: <AANLkTim8UvXmpx0udG-krpkdp9zDrtt5Gk-InKp2AjBF@mail.gmail.com>
References: <1427456515.20100526174916@eltc.ru>
	<C7E0DD6C-0EBB-4C1F-A47A-D3685DF5C6FC@jerris.com>
	<1751544061.20100527132304@eltc.ru>
	<AANLkTimUWPk_SoDoIRHbkfISJz0ibNfyWLulZxsn9gQ0@mail.gmail.com>
	<141728530.20100528013547@eltc.ru>
	<AANLkTinpkGvmvUvOQzAQdmUHHMNN_kxWQY304wlMEBwC@mail.gmail.com>
	<55492769.20100528021228@eltc.ru>
	<AANLkTim8UvXmpx0udG-krpkdp9zDrtt5Gk-InKp2AjBF@mail.gmail.com>
Message-ID: <262607082.20100528093416@eltc.ru>

Hi, Anthony.

You wrote 28 may 2010, 3:50:37:

> not sure what you mean but i'm simply explaining to you that you cannot
> re-invite to change the media in the SOA before the initial call was setup.

Big thanks Anthony for the consultation!

Regards
Sergey Scheglov/





From codeghar at gmail.com  Thu May 27 20:34:23 2010
From: codeghar at gmail.com (Code Ghar)
Date: Thu, 27 May 2010 22:34:23 -0500
Subject: [Freeswitch-users] FS as Media Gateway Only
In-Reply-To: <AANLkTikoxsXFuYBxsXXtB26Y9_8f1I1wGmE-u-I_l9PP@mail.gmail.com>
References: <AANLkTimgKRtL06xmyjXUPnpNIjlYkHC6BytzmTqKmVm7@mail.gmail.com>
	<AANLkTikoxsXFuYBxsXXtB26Y9_8f1I1wGmE-u-I_l9PP@mail.gmail.com>
Message-ID: <AANLkTikgol1TigK9G9vvk90ogtjqeJ3nwPXqdFUXcV5W@mail.gmail.com>

Hi Vitalie

Thanks for providing the link and details. If I understood correctly, the
chain of signaling would be Inbound -> FSSIP -> FSRTP -> Outbound (using
names and terms in my original question), while the chain of media would be
Inbound -> FSRTP -> Outbound. This way we can have multiple servers handling
media and minimal servers handling signaling.

Let me clarify a little more my motivation for asking this question in the
first place. I work with telecom carriers on a daily basis and have seen
many different architectures. The first biggest problem is how to load
balance SIP traffic when you are receiving calls, if one server is not
enough. The second biggest problem is handling all RTP, including
transcoding. With this architecture, one or two IPs for signaling can be
handled by most carriers. So you can beef up your hardware for signaling and
depend less on your carrier's ability to load balance traffic for you. If
they can do round-robin or failover for two IPs, you are golden. And then
you can deploy multiple nodes to handle all RTP duties, without having to
concern your carrier about multiple servers and IPs. But there's one thing
still missing. Your outbound carrier still needs to allow traffic from
multiple IPs because now they are dealing with FSRTP instead of FSSIP.

Is there such a solution possible using FS that all signaling, for both
inbound and outbound carriers, can be handled by a couple of FSSIP nodes
(depending on the amount of traffic, maybe a few more) while delegating
media responsibilities to multiple FSRTP nodes? In this situation, SIP IP is
always, say 10.10.10.1 or 10.10.10.2, but the SDP may use media IPs
10.10.10.3, 10.10.10.4, 10.10.10.5, and so on. Almost all carriers I have
seen concern themselves only with which SIP IPs they should allow and don't
care how many or which media IPs they have to deal with. Any ideas on
minimizing signaling IPs while adding more media IPs as required?

I have seen re-invite being used in production where you can just let your
inbound and outbound handle media between them on their own without it going
through your network. But there are circumstances where people might need to
pass media through their own network, chiefly to perform transcoding and
also to handle other interop issues. It is because of this use case,
combined with the need for minimizing signaling IPs, that the original
question was asked.




On Thu, May 27, 2010 at 1:15 AM, Vitalii Colosov <vetali100 at gmail.com>wrote:

> Hi Code,
>
> I have working example doing exactly what you've described.
> One signalling FS bridges incoming call to a set of media servers
> (depending on ip, but you can implement any routing logic including round
> robin) and then transfers media stream after bridging to that media server.
>
> You can achieve this on signalling FS by creating a Lua script that
> contains the following lines:
>
> media_server="my_media_X.mydomain.com"; --to be determined by routing
> logic
> forwarding_session = "sofia/external/"..called_number.."@"..media_server;
> session:setVariable("bypass_media_after_bridge", "true");
> session:setVariable("hangup_after_bridge", "true");
> session:execute("bridge",forwarding_session);
>
> The call will arrive to the selected media server successfully and media
> stream will start bypassing signalling FS after bridge.
>
> You can read the following thread, it describes how you can setup such
> configuration.
>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-March/055231.html
>
> I think it will fit your needs.
>
> Regards,
> Vitalie
>
>
> 2010/5/27 Code Ghar <codeghar at gmail.com>
>
>> Is it possible -- and are there any case studies, practical experience,
>> etc -- on deploying FreeSWITCH (FS) in this architecture: one server (FSSIP)
>> handles SIP signaling only, and multiple servers (FSRTP1, FSRTP2, ...,
>> FSRTPn) handle all media responsibilities? So when a call comes in, the SDP
>> contains IP of, say FSRTP1, as media handler. For this to work, FSSIP would
>> request FSRTPx for media resources for each new call and add its IP and port
>> in SDP. The media servers/gateways would play IVR, etc.; collect DTMF and
>> forward as appropriate to FSSIP; perform transcoding; etc.; all while FSSIP
>> only deals with signaling. This way multiple servers could be deployed to
>> handle media responsibilities and only a handful would be required for
>> signaling. In future if there's a greater need for transcoding, etc. all you
>> need to do is deploy a media server and not have to add servers for
>> signaling.
>>
>> This idea came to me because I have come across two proprietary
>> applications that do it this way. They have a SIP component and a media
>> component. You can run both on the same physical machine or you can separate
>> them out into multiple machines.
>>
>> Another way for this could be to integrate FS as a media component to
>> another application's SIP component. A mix-and-match, so to speak.
>>
>> On the flip side, deploy FS as a SIP server and use media capabilities of
>> some other hardware or software application. For example, FS handles
>> signaling and use dedicated hardware for media. A good example of this is
>> illustrated (somewhat) by an image on Sangoma's website:
>> http://www.sangoma.com/assets/images/content/transcoding_diagram.jpg.
>> Look at the "pooled transcoding".
>>
>> Is FS even designed to be this modular? If so, how can the aforementioned
>> scenario(s) be achieved?
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jmesquita at freeswitch.org  Thu May 27 21:49:53 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 28 May 2010 01:49:53 -0300
Subject: [Freeswitch-users] FSComm linker issue
In-Reply-To: <AANLkTiliaONCmhMhVsgiXi05v5dIwASq7Zb2t-QJf1YX@mail.gmail.com>
References: <AANLkTiliaONCmhMhVsgiXi05v5dIwASq7Zb2t-QJf1YX@mail.gmail.com>
Message-ID: <AANLkTimnGnnxiWW2DmMTujlOHalpjdHhOOZTMxx_DAB2@mail.gmail.com>

I must have missed that other email, I am sorry about that. The problem is
that the VS Proj was not update with the latest code modifications I've
made. I am the sole developer of FSComm and I develop/test in Mac/Linux.
It's quite hard for me to keep up with all platforms by myself. Do you feel
you could contribute with your MS knowledge? If you feel you can, I can
provide you the pointers of what files were added and let you know.

Otherwise, could you open a Jira so I won't forget to get to that when I
have the time?

Thank you for your interest on the project.

Regards,
Jo?o Mesquita


On Thu, May 27, 2010 at 7:35 AM, Jingwei Yang <jingwei.yang at gmail.com>wrote:

> Hello,
>
> I encountered six linker errors when compiling the latest codes of FSComm
> in VS2008. I followed the steps from here:
> http://wiki.freeswitch.org/wiki/FSComm#Windows. Someone else seemed to
> have the same issue but got no answers:
> http://article.gmane.org/gmane.comp.telephony.freeswitch.user/21757/match=fscomm
> .
>
> Please enlighten me how to deal with these errors.
>
> Thanks and best regards,
> -Jingwei
>
> 10>fshost.obj : error LNK2019: unresolved external symbol "public:
> __thiscall Channel::Channel(class QString)" (??0Channel@@QAE at VQString@@@Z)
> referenced in function "private: void __thiscall
> FSHost::eventChannelCreate(class QSharedPointer<struct switch_event>,class
> QString)" (?eventChannelCreate at FSHost@@AAEXV?$QSharedPointer at Uswitch_event
> @@@@VQString@@@Z)
>
> 10>mainwindow.obj : error LNK2019: unresolved external symbol "public:
> __thiscall StateDebugDialog::StateDebugDialog(class QWidget *)"
> (??0StateDebugDialog@@QAE at PAVQWidget@@@Z) referenced in function "private:
> void __thiscall MainWindow::debugEventsTriggered(void)"
> (?debugEventsTriggered at MainWindow@@AAEXXZ)
>
> 10>mainwindow.obj : error LNK2019: unresolved external symbol "public:
> __thiscall ConsoleWindow::ConsoleWindow(class QWidget *)" (??0ConsoleWindow@
> @QAE at PAVQWidget@@@Z) referenced in function "private: void __thiscall
> MainWindow::debugConsoleTriggered(void)" (?debugConsoleTriggered at MainWindow
> @@AAEXXZ)
>
> 10>prefdialog.obj : error LNK2019: unresolved external symbol "public:
> __thiscall CodecWidget::CodecWidget(class QWidget *)" (??0CodecWidget@
> @QAE at PAVQWidget@@@Z) referenced in function "public: void __thiscall
> Ui_PrefDialog::setupUi(class QDialog *)" (?setupUi at Ui_PrefDialog
> @@QAEXPAVQDialog@@@Z)
>
> 10>prefsofia.obj : error LNK2019: unresolved external symbol "public: void
> __thiscall CodecWidget::setCodecString(class QString)"
> (?setCodecString at CodecWidget@@QAEXVQString@@@Z) referenced in function
> "public: void __thiscall PrefSofia::readConfig(void)" (?readConfig at PrefSofia
> @@QAEXXZ)
>
> 10>prefsofia.obj : error LNK2019: unresolved external symbol "public: class
> QString __thiscall CodecWidget::getCodecString(void)"
> (?getCodecString at CodecWidget@@QAE?AVQString@@XZ) referenced in function
> "public: void __thiscall PrefSofia::writeConfig(void)"
> (?writeConfig at PrefSofia@@QAEXXZ)
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at gmail.com  Thu May 27 23:06:23 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Fri, 28 May 2010 08:06:23 +0200
Subject: [Freeswitch-users] 407 Proxy Authentication
In-Reply-To: <AANLkTikJ9zW_yH0hH0GrVK5ovFbcbY9XOie-54h55f4P@mail.gmail.com>
References: <AANLkTikJ9zW_yH0hH0GrVK5ovFbcbY9XOie-54h55f4P@mail.gmail.com>
Message-ID: <3015721B-23C7-4174-839A-84DB8DFB7379@gmail.com>

If 63.79.178.192 is Verizon, It's Verizon sending the 407, so asking  
you to authenticate.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 28/05/2010 ? 03:19, Michael De Lorenzo a ?crit :

> I've tried updating our ACL conf, but that doesn't seem to help as I  
> still get the 407 error (line 50 from the Gist below).  Am I correct  
> that the authentication is failing when Verizon attempts to contact  
> our switch? Or is it an authentication failure when I hit Verizon?
>
> Here's the trace of the call attempt (I've replaced an IP address  
> and phone number, so I do realize that they're not correct) - http://gist.github.com/416604
>
> send 1129 bytes to udp/[63.79.178.192]:5060 at 01:04:26.090037:
>     
> ------------------------------------------------------------------------
>    INVITE sip:17895551234 at 63.79.178.192 SIP/2.0
>    Via: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK0NZjmNS5Byj4c
>    Max-Forwards: 70
>    From: "C3 Mgmt" <sip: 
> 9727289377 at 63.79.178.192;transport=udp>;tag=tgHN0ScKyNcXr
>    To: <sip:17895551234 at 63.79.178.192>
>    Call-ID: cd26b853-e497-122d-278a-000bdb94aab9
>    CSeq: 131385997 INVITE
>    Contact: <sip:gw+verizon at 0.0.0.0:5080;transport=udp;gw=verizon>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-28484a1 2010-04-22  
> 15:06:05 -0400
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,  
> REGISTER, REFER, NOTIFY
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, hold, refer
>    Privacy: none
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 291
>    X-FS-Support: update_display
>    P-Asserted-Identity: "C3 Mgmt" <sip:+19727289377 at 63.79.178.192>
>
>    v=0
>    o=FreeSWITCH 1274985280 1274985281 IN IP4 0.0.0.0
>    s=FreeSWITCH
>    c=IN IP4 0.0.0.0
>    t=0 0
>    m=audio 21600 RTP/AVP 0 8 3 101 13
>    a=rtpmap:0 PCMU/8000
>    a=rtpmap:8 PCMA/8000
>    a=rtpmap:3 GSM/8000
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=rtpmap:13 CN/8000
>    a=ptime:20
>     
> ------------------------------------------------------------------------
> 2010-05-27 21:00:10.128406 [DEBUG] sofia.c:4168 Channel sofia/ 
> external/17895551234 entering state [calling][0]
> recv 296 bytes from udp/[63.79.178.192]:5060 at 01:00:10.192551:
>     
> ------------------------------------------------------------------------
>    SIP/2.0 100 Trying
>    v: SIP/2.0/UDP  
> 0.0.0.0:5080;rport;branch=z9hG4bKZc6Sjt81eNvHH;received=0.0.0.0
>    f: "C3 Mgmt" <sip: 
> 9727289377 at 63.79.178.192;transport=udp>;tag=S7QvyyUF1cpaD
>    t: <sip:17895551234 at 63.79.178.192>
>    i: 348b8efb-e497-122d-278a-000bdb94aab9
>    CSeq: 131385870 INVITE
>    l: 0
>
>     
> ------------------------------------------------------------------------
> recv 425 bytes from udp/[63.79.178.192]:5060 at 01:00:10.192710:
>     
> ------------------------------------------------------------------------
>    SIP/2.0 407 Proxy Authentication Required
>    v: SIP/2.0/UDP  
> 0.0.0.0:5080;rport;branch=z9hG4bKZc6Sjt81eNvHH;received=0.0.0.0
>    f: "C3 Mgmt" <sip: 
> 9727289377 at 63.79.178.192;transport=udp>;tag=S7QvyyUF1cpaD
>    t: <sip:17895551234 at 63.79.178.192>;tag=22703bab
>    i: 348b8efb-e497-122d-278a-000bdb94aab9
>    CSeq: 131385870 INVITE
>    l: 0
>    Proxy-Authenticate: DIGEST  
> realm="WCOM",nonce="a586274e395fb9b6f66f1a7829b4531a.1275008409"
>
>     
> ------------------------------------------------------------------------
> send 350 bytes to udp/[63.79.178.192]:5060 at 01:00:10.192977:
>     
> ------------------------------------------------------------------------
>    ACK sip:17895551234 at 63.79.178.192 SIP/2.0
>    Via: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bKZc6Sjt81eNvHH
>    Max-Forwards: 70
>    From: "C3 Mgmt" <sip: 
> 9727289377 at 63.79.178.192;transport=udp>;tag=S7QvyyUF1cpaD
>    To: <sip:17895551234 at 63.79.178.192>;tag=22703bab
>    Call-ID: 348b8efb-e497-122d-278a-000bdb94aab9
>    CSeq: 131385870 ACK
>    Content-Length: 0
>
>     
> ------------------------------------------------------------------------
> 2010-05-27 21:00:10.193015 [DEBUG] sofia.c:4168 Channel sofia/ 
> external/17895551234 entering state [terminated][904]
> 2010-05-27 21:00:10.193015 [NOTICE] sofia.c:4804 Hangup sofia/ 
> external/17895551234 [CS_CONSUME_MEDIA] [NORMAL_UNSPECIFIED]
> 2010-05-27 21:00:10.193015 [DEBUG] switch_channel.c:2117 Send signal  
> sofia/external/17895551234 [KILL]
> 2010-05-27 21:00:10.193015 [DEBUG] switch_core_session.c:1022 Send  
> signal sofia/external/17895551234 [BREAK]
> 2010-05-27 21:00:10.193015 [DEBUG] switch_core_state_machine.c:314  
> (sofia/external/17895551234) Running State Change CS_HANGUP
> 2010-05-27 21:00:10.193015 [DEBUG] switch_core_state_machine.c:499  
> (sofia/external/17895551234) State HANGUP
> 2010-05-27 21:00:10.193015 [DEBUG] mod_sofia.c:410 sofia/external/ 
> 17895551234 Overriding SIP cause 480 with 904 from the other leg
> 2010-05-27 21:00:10.193015 [DEBUG] mod_sofia.c:416 Channel sofia/ 
> external/17895551234 hanging up, cause: NORMAL_UNSPECIFIED
> 2010-05-27 21:00:10.193015 [DEBUG] switch_core_state_machine.c:46  
> sofia/external/17895551234 Standard HANGUP, cause: NORMAL_UNSPECIFIED
> 2010-05-27 21:00:10.193015 [DEBUG] switch_core_state_machine.c:499  
> (sofia/external/17895551234) State HANGUP going to sleep
> 2010-05-27 21:00:10.193015 [DEBUG] switch_core_state_machine.c:333  
> (sofia/external/17895551234) State Change CS_HANGUP -> CS_REPORTING
> 2010-05-27 21:00:10.193015 [DEBUG] switch_core_session.c:1022 Send  
> signal sofia/external/17895551234 [BREAK]
> 2010-05-27 21:00:10.193015 [DEBUG] switch_core_state_machine.c:314  
> (sofia/external/17895551234) Running State Change CS_REPORTING
> 2010-05-27 21:00:10.193015 [DEBUG] switch_ivr_originate.c:3228  
> Originate Resulted in Error Cause: 31 [NORMAL_UNSPECIFIED]
> 2010-05-27 21:00:10.193015 [DEBUG] switch_core_state_machine.c:590  
> (sofia/external/17895551234) State REPORTING
>
>
> freeswitch at govinteract-fs-dev-2> 2010-05-27 21:00:10.299296 [DEBUG]  
> switch_core_state_machine.c:53 sofia/external/17895551234 Standard  
> REPORTING, cause: NORMAL_UNSPECIFIED
> 2010-05-27 21:00:10.299296 [DEBUG] switch_core_state_machine.c:590  
> (sofia/external/17895551234) State REPORTING going to sleep
> 2010-05-27 21:00:10.299296 [DEBUG] switch_core_state_machine.c:327  
> (sofia/external/17895551234) State Change CS_REPORTING -> CS_DESTROY
> 2010-05-27 21:00:10.299296 [DEBUG] switch_core_session.c:1022 Send  
> signal sofia/external/17895551234 [BREAK]
> 2010-05-27 21:00:10.299296 [DEBUG] switch_core_session.c:1165  
> Session 6 (sofia/external/17895551234) Locked, Waiting on external  
> entities
> 2010-05-27 21:00:10.299296 [NOTICE] switch_core_session.c:1183  
> Session 6 (sofia/external/17895551234) Ended
> 2010-05-27 21:00:10.299296 [NOTICE] switch_core_session.c:1185 Close  
> Channel sofia/external/17895551234 [CS_DESTROY]
> 2010-05-27 21:00:10.299296 [DEBUG] switch_core_state_machine.c:428  
> (sofia/external/17895551234) Running State Change CS_DESTROY
> 2010-05-27 21:00:10.299296 [DEBUG] switch_core_state_machine.c:439  
> (sofia/external/17895551234) State DESTROY
> 2010-05-27 21:00:10.299296 [DEBUG] mod_sofia.c:343 sofia/external/ 
> 17895551234 SOFIA DESTROY
> 2010-05-27 21:00:10.299296 [DEBUG] switch_core_state_machine.c:60  
> sofia/external/17895551234 Standard DESTROY
> 2010-05-27 21:00:10.299296 [DEBUG] switch_core_state_machine.c:439  
> (sofia/external/17895551234) State DESTROY going to sleep
>
> freeswitch at govinteract-fs-dev-2> lua test1.lua
> 2010-05-27 21:04:26.087322 [DEBUG] switch_ivr_originate.c:1885  
> variable string 0 = [sip_cid_type=pid]
> 2010-05-27 21:04:26.087322 [DEBUG] switch_ivr_originate.c:1885  
> variable string 1 = [origination_caller_id_name=C3 Mgmt]
> 2010-05-27 21:04:26.087322 [DEBUG] switch_ivr_originate.c:1885  
> variable string 2 = [origination_caller_id_number=+19727289377]
> 2010-05-27 21:04:26.087322 [DEBUG] switch_ivr_originate.c:1885  
> variable string 3 = [ignore_early_media=true]
> 2010-05-27 21:04:26.088360 [NOTICE] switch_channel.c:669 New Channel  
> sofia/external/17895551234 [f5c8454e-69f4-11df-b69d-099cfc924087]
> 2010-05-27 21:04:26.088360 [DEBUG] mod_sofia.c:3444 (sofia/external/ 
> 17895551234) State Change CS_NEW -> CS_INIT
> 2010-05-27 21:04:26.088360 [DEBUG] switch_core_session.c:1022 Send  
> signal sofia/external/17895551234 [BREAK]
> 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:314  
> (sofia/external/17895551234) Running State Change CS_INIT
> 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:338  
> (sofia/external/17895551234) State INIT
> 2010-05-27 21:04:26.088360 [DEBUG] mod_sofia.c:83 sofia/external/ 
> 17895551234 SOFIA INIT
> 2010-05-27 21:04:26.088360 [DEBUG] mod_sofia.c:117 (sofia/external/ 
> 17895551234) State Change CS_INIT -> CS_ROUTING
> 2010-05-27 21:04:26.088360 [DEBUG] switch_core_session.c:1022 Send  
> signal sofia/external/17895551234 [BREAK]
> 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:338  
> (sofia/external/17895551234) State INIT going to sleep
> 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:314  
> (sofia/external/17895551234) Running State Change CS_ROUTING
> 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:341  
> (sofia/external/17895551234) State ROUTING
> 2010-05-27 21:04:26.088360 [DEBUG] mod_sofia.c:140 sofia/external/ 
> 17895551234 SOFIA ROUTING
> 2010-05-27 21:04:26.088360 [DEBUG] switch_ivr_originate.c:66 (sofia/ 
> external/17895551234) State Change CS_ROUTING -> CS_CONSUME_MEDIA
> 2010-05-27 21:04:26.088360 [DEBUG] switch_core_session.c:1022 Send  
> signal sofia/external/17895551234 [BREAK]
> 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:341  
> (sofia/external/17895551234) State ROUTING going to sleep
> 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:314  
> (sofia/external/17895551234) Running State Change CS_CONSUME_MEDIA
> 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:360  
> (sofia/external/17895551234) State CONSUME_MEDIA
> 2010-05-27 21:04:26.088360 [DEBUG] switch_core_state_machine.c:360  
> (sofia/external/17895551234) State CONSUME_MEDIA going to sleep
> send 1129 bytes to udp/[63.79.178.192]:5060 at 01:04:26.090037:
>     
> ------------------------------------------------------------------------
>    INVITE sip:17895551234 at 63.79.178.192 SIP/2.0
>    Via: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK0NZjmNS5Byj4c
>    Max-Forwards: 70
>    From: "C3 Mgmt" <sip: 
> 9727289377 at 63.79.178.192;transport=udp>;tag=tgHN0ScKyNcXr
>    To: <sip:17895551234 at 63.79.178.192>
>    Call-ID: cd26b853-e497-122d-278a-000bdb94aab9
>    CSeq: 131385997 INVITE
>    Contact: <sip:gw+verizon at 0.0.0.0:5080;transport=udp;gw=verizon>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-28484a1 2010-04-22  
> 15:06:05 -0400
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,  
> REGISTER, REFER, NOTIFY
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, hold, refer
>    Privacy: none
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 291
>    X-FS-Support: update_display
>    P-Asserted-Identity: "C3 Mgmt" <sip:+19727289377 at 63.79.178.192>
>
>    v=0
>    o=FreeSWITCH 1274985280 1274985281 IN IP4 0.0.0.0
>    s=FreeSWITCH
>    c=IN IP4 0.0.0.0
>    t=0 0
>    m=audio 23386 RTP/AVP 0 8 3 101 13
>    a=rtpmap:0 PCMU/8000
>    a=rtpmap:8 PCMA/8000
>    a=rtpmap:3 GSM/8000
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=rtpmap:13 CN/8000
>    a=ptime:20
>     
> ------------------------------------------------------------------------
> 2010-05-27 21:04:26.089756 [DEBUG] sofia.c:4168 Channel sofia/ 
> external/17895551234 entering state [calling][0]
> recv 425 bytes from udp/[63.79.178.192]:5060 at 01:04:26.167603:
>     
> ------------------------------------------------------------------------
>    SIP/2.0 407 Proxy Authentication Required
>    v: SIP/2.0/UDP  
> 0.0.0.0:5080;rport;branch=z9hG4bK0NZjmNS5Byj4c;received=0.0.0.0
>    f: "C3 Mgmt" <sip: 
> 9727289377 at 63.79.178.192;transport=udp>;tag=tgHN0ScKyNcXr
>    t: <sip:17895551234 at 63.79.178.192>;tag=41e44fd5
>    i: cd26b853-e497-122d-278a-000bdb94aab9
>    CSeq: 131385997 INVITE
>    l: 0
>    Proxy-Authenticate: DIGEST  
> realm="WCOM",nonce="43ea850af91d419e72d85c0375a60237.1275008665"
>
>     
> ------------------------------------------------------------------------
> send 350 bytes to udp/[63.79.178.192]:5060 at 01:04:26.167913:
>     
> ------------------------------------------------------------------------
>    ACK sip:17895551234 at 63.79.178.192 SIP/2.0
>    Via: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK0NZjmNS5Byj4c
>    Max-Forwards: 70
>    From: "C3 Mgmt" <sip: 
> 9727289377 at 63.79.178.192;transport=udp>;tag=tgHN0ScKyNcXr
>    To: <sip:17895551234 at 63.79.178.192>;tag=41e44fd5
>    Call-ID: cd26b853-e497-122d-278a-000bdb94aab9
>    CSeq: 131385997 ACK
>    Content-Length: 0
>
>     
> ------------------------------------------------------------------------
> send 1359 bytes to udp/[63.79.178.192]:5060 at 01:04:26.168477:
>     
> ------------------------------------------------------------------------
>    INVITE sip:17895551234 at 63.79.178.192 SIP/2.0
>    Via: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK1yrBpga9868pr
>    Max-Forwards: 70
>    From: "C3 Mgmt" <sip: 
> 9727289377 at 63.79.178.192;transport=udp>;tag=tgHN0ScKyNcXr
>    To: <sip:17895551234 at 63.79.178.192>
>    Call-ID: cd26b853-e497-122d-278a-000bdb94aab9
>    CSeq: 131385998 INVITE
>    Contact: <sip:gw+verizon at 0.0.0.0:5080;transport=udp;gw=verizon>
>    Expires: 3600
>    User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-28484a1 2010-04-22  
> 15:06:05 -0400
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,  
> REGISTER, REFER, NOTIFY
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, hold, refer
>    Proxy-Authorization: Digest username="9727289377", realm="WCOM",  
> nonce="43ea850af91d419e72d85c0375a60237.1275008665", algorithm=MD5,  
> uri="sip:17895551234 at 63.79.178.192",  
> response="1871a9dbd248f92b65df2af84c6057f8"
>    Privacy: none
>    Content-Type: application/sdp
>    Content-Disposition: session
>    Content-Length: 291
>    X-FS-Support: update_display
>    P-Asserted-Identity: "C3 Mgmt" <sip:+19727289377 at 63.79.178.192>
>
>    v=0
>    o=FreeSWITCH 1274985280 1274985281 IN IP4 0.0.0.0
>    s=FreeSWITCH
>    c=IN IP4 0.0.0.0
>    t=0 0
>    m=audio 23386 RTP/AVP 0 8 3 101 13
>    a=rtpmap:0 PCMU/8000
>    a=rtpmap:8 PCMA/8000
>    a=rtpmap:3 GSM/8000
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=rtpmap:13 CN/8000
>    a=ptime:20
>     
> ------------------------------------------------------------------------
> 2010-05-27 21:04:26.168049 [DEBUG] sofia.c:4168 Channel sofia/ 
> external/17895551234 entering state [calling][0]
> recv 296 bytes from udp/[63.79.178.192]:5060 at 01:04:26.231231:
>     
> ------------------------------------------------------------------------
>    SIP/2.0 100 Trying
>    v: SIP/2.0/UDP  
> 0.0.0.0:5080;rport;branch=z9hG4bK1yrBpga9868pr;received=0.0.0.0
>    f: "C3 Mgmt" <sip: 
> 9727289377 at 63.79.178.192;transport=udp>;tag=tgHN0ScKyNcXr
>    t: <sip:17895551234 at 63.79.178.192>
>    i: cd26b853-e497-122d-278a-000bdb94aab9
>    CSeq: 131385998 INVITE
>    l: 0
>
>     
> ------------------------------------------------------------------------
> recv 425 bytes from udp/[63.79.178.192]:5060 at 01:04:26.231408:
>     
> ------------------------------------------------------------------------
>    SIP/2.0 407 Proxy Authentication Required
>    v: SIP/2.0/UDP  
> 0.0.0.0:5080;rport;branch=z9hG4bK1yrBpga9868pr;received=0.0.0.0
>    f: "C3 Mgmt" <sip: 
> 9727289377 at 63.79.178.192;transport=udp>;tag=tgHN0ScKyNcXr
>    t: <sip:17895551234 at 63.79.178.192>;tag=1da20439
>    i: cd26b853-e497-122d-278a-000bdb94aab9
>    CSeq: 131385998 INVITE
>    l: 0
>    Proxy-Authenticate: DIGEST  
> realm="WCOM",nonce="43ea850af91d419e72d85c0375a60237.1275008665"
>
>     
> ------------------------------------------------------------------------
> send 350 bytes to udp/[63.79.178.192]:5060 at 01:04:26.231667:
>     
> ------------------------------------------------------------------------
>    ACK sip:17895551234 at 63.79.178.192 SIP/2.0
>    Via: SIP/2.0/UDP 0.0.0.0:5080;rport;branch=z9hG4bK1yrBpga9868pr
>    Max-Forwards: 70
>    From: "C3 Mgmt" <sip: 
> 9727289377 at 63.79.178.192;transport=udp>;tag=tgHN0ScKyNcXr
>    To: <sip:17895551234 at 63.79.178.192>;tag=1da20439
>    Call-ID: cd26b853-e497-122d-278a-000bdb94aab9
>    CSeq: 131385998 ACK
>    Content-Length: 0
>
>     
> ------------------------------------------------------------------------
> 2010-05-27 21:04:26.231294 [DEBUG] sofia.c:4168 Channel sofia/ 
> external/17895551234 entering state [terminated][904]
> 2010-05-27 21:04:26.231294 [NOTICE] sofia.c:4804 Hangup sofia/ 
> external/17895551234 [CS_CONSUME_MEDIA] [NORMAL_UNSPECIFIED]
> 2010-05-27 21:04:26.231294 [DEBUG] switch_channel.c:2117 Send signal  
> sofia/external/17895551234 [KILL]
> 2010-05-27 21:04:26.231294 [DEBUG] switch_core_session.c:1022 Send  
> signal sofia/external/17895551234 [BREAK]
> 2010-05-27 21:04:26.232334 [DEBUG] switch_core_state_machine.c:314  
> (sofia/external/17895551234) Running State Change CS_HANGUP
> 2010-05-27 21:04:26.232334 [DEBUG] switch_core_state_machine.c:499  
> (sofia/external/17895551234) State HANGUP
> 2010-05-27 21:04:26.232334 [DEBUG] mod_sofia.c:410 sofia/external/ 
> 17895551234 Overriding SIP cause 480 with 904 from the other leg
> 2010-05-27 21:04:26.232334 [DEBUG] mod_sofia.c:416 Channel sofia/ 
> external/17895551234 hanging up, cause: NORMAL_UNSPECIFIED
> 2010-05-27 21:04:26.232334 [DEBUG] switch_core_state_machine.c:46  
> sofia/external/17895551234 Standard HANGUP, cause: NORMAL_UNSPECIFIED
> 2010-05-27 21:04:26.232334 [DEBUG] switch_core_state_machine.c:499  
> (sofia/external/17895551234) State HANGUP going to sleep
> 2010-05-27 21:04:26.232334 [DEBUG] switch_core_state_machine.c:333  
> (sofia/external/17895551234) State Change CS_HANGUP -> CS_REPORTING
> 2010-05-27 21:04:26.232334 [DEBUG] switch_core_session.c:1022 Send  
> signal sofia/external/17895551234 [BREAK]
> 2010-05-27 21:04:26.232334 [DEBUG] switch_core_state_machine.c:314  
> (sofia/external/17895551234) Running State Change CS_REPORTING
> 2010-05-27 21:04:26.232334 [DEBUG] switch_ivr_originate.c:3228  
> Originate Resulted in Error Cause: 31 [NORMAL_UNSPECIFIED]
> 2010-05-27 21:04:26.232334 [DEBUG] switch_core_state_machine.c:590  
> (sofia/external/17895551234) State REPORTING
>
>
> freeswitch at govinteract-fs-dev-2> 2010-05-27 21:04:26.260365 [DEBUG]  
> switch_core_state_machine.c:53 sofia/external/17895551234 Standard  
> REPORTING, cause: NORMAL_UNSPECIFIED
> 2010-05-27 21:04:26.260365 [DEBUG] switch_core_state_machine.c:590  
> (sofia/external/17895551234) State REPORTING going to sleep
> 2010-05-27 21:04:26.260365 [DEBUG] switch_core_state_machine.c:327  
> (sofia/external/17895551234) State Change CS_REPORTING -> CS_DESTROY
> 2010-05-27 21:04:26.260365 [DEBUG] switch_core_session.c:1022 Send  
> signal sofia/external/17895551234 [BREAK]
> 2010-05-27 21:04:26.260365 [DEBUG] switch_core_session.c:1165  
> Session 7 (sofia/external/17895551234) Locked, Waiting on external  
> entities
> 2010-05-27 21:04:26.260365 [NOTICE] switch_core_session.c:1183  
> Session 7 (sofia/external/17895551234) Ended
> 2010-05-27 21:04:26.260365 [NOTICE] switch_core_session.c:1185 Close  
> Channel sofia/external/17895551234 [CS_DESTROY]
> 2010-05-27 21:04:26.260365 [DEBUG] switch_core_state_machine.c:428  
> (sofia/external/17895551234) Running State Change CS_DESTROY
> 2010-05-27 21:04:26.260365 [DEBUG] switch_core_state_machine.c:439  
> (sofia/external/17895551234) State DESTROY
> 2010-05-27 21:04:26.260365 [DEBUG] mod_sofia.c:343 sofia/external/ 
> 17895551234 SOFIA DESTROY
> 2010-05-27 21:04:26.260365 [DEBUG] switch_core_state_machine.c:60  
> sofia/external/17895551234 Standard DESTROY
> 2010-05-27 21:04:26.260365 [DEBUG] switch_core_state_machine.c:439  
> (sofia/external/17895551234) State DESTROY going to sleep
> Here's how I've edited our acl.conf.xml file:
>
> <configuration name="acl.conf" description="Network Lists">
>   <network-lists>
>     <!--
>          These ACL's are automatically created on startup.
>
>          rfc1918.auto  - RFC1918 Space
>          nat.auto      - RFC1918 Excluding your local lan.
>          localnet.auto - ACL for your local lan.
>          loopback.auto - ACL for your local lan.
>     -->
>
>     <list name="lan" default="allow">
>       <node type="deny" cidr="192.168.42.0/24"/>
>       <node type="allow" cidr="192.168.42.42/32"/>
>    </list>
>    <list name="verizon" default="deny">
>       <node type="allow" cidr="63.79.178.192"/>
>    </list>
>    .... truncated, the rest is the default file ....
>
> Any help would be greatly appreciated!
>
>
> 2010/5/27 <freeswitch-users-request at lists.freeswitch.org>
> Send FreeSWITCH-users mailing list submissions to
>        freeswitch-users at lists.freeswitch.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
>        http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> or, via email, send a message with subject or body 'help' to
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>
> You can reach the person managing the list at
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>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of FreeSWITCH-users digest..."
>
> Today's Topics:
>
>   1. Re: Don't work playback after bypass media mode. (Sergey  
> Scheglov)
>   2. Re: Don't work playback after bypass media mode. (Sergey  
> Scheglov)
>   3. Re: Don't work playback after bypass media mode.
>      (Anthony Minessale)
>   4. Re: Don't work playback after bypass media mode. (Sergey  
> Scheglov)
>   5. Re: Don't work playback after bypass media mode. (Sergey  
> Scheglov)
>
>
> ---------- Forwarded message ----------
> From: Sergey Scheglov <sid at eltc.ru>
> To: freeswitch-users at lists.freeswitch.org
> Date: Fri, 28 May 2010 01:31:26 +0700
> Subject: Re: [Freeswitch-users] Don't work playback after bypass  
> media mode.
> Hi, Anthony.
>
>
>
> Thank's for reply.
>
>
>
> You wrote 27 may 2010 ?., 20:34:15:
>
>
>
> > playback executes on line 87 of your trace.
>
>
>
> yes, but executed by log after 30 seconds after line 84 and  
> immediately hangup.
>
> Note, duration wav file - 4 sec.
>
>
>
> > If you do not hear the audio, It means the re-establishment of  
> media fails somehow based on your topology or
>
> the phone you are on does not support early media,
>
>
>
> I don't  hear the audio, because FS don't send RTP traffic to my  
> phone in early media mode (checked sniffers) in my case.
>
> If dialplans is:
>
> <extension name="fail_balance">
>
>       <condition field="${blabla}" expression="^\-1$">
>
>         <action application="pre_answer"/>
>
>         <action application="sleep" data="1000"/>
>
>         <action application="playback" data="elight/neg_balance.wav"/>
>
>         <action application="hangup" data="BEARERCAPABILITY_NOTAUTH"/>
>
>       </condition>
>
>  </extension>
>
> then work's fine, no problems (bypass_media=true not set).
>
>
>
> > change your pre-answer before the playback to answer to verify.
>
>
>
> If set answer, problem disappears. But answer it's not for my case.
>
>
>
> > Try adding sofia profile internal siptrace on to see the sip  
> traffic too.
>
>
>
> Call log with sip trace http://pastebin.freeswitch.org/13065
>
>
>
> Thanks again :)
>
>
>
> -- 
>
> Regard's
>
> Sergey Scheglov
>
>
>
> ---------- Forwarded message ----------
> From: Sergey Scheglov <sid at eltc.ru>
> To: Anthony Minessale <freeswitch-users at lists.freeswitch.org>
> Date: Fri, 28 May 2010 01:35:47 +0700
> Subject: Re: [Freeswitch-users] Don't work playback after bypass  
> media mode.
> Hi, Anthony.
>
>
>
> You wrote 27 may 2010 ?., 20:34:15:
>
>
>
>
> playback executes on line 87 of your trace.
>
>
>
> If you do not hear the audio, It means the re-establishment of media  
> fails somehow based on your topology or
>
> the phone you are on does not support early media, change your pre- 
> answer before the playback to answer to verify.
>
>
>
> Try adding sofia profile internal siptrace on to see the sip traffic  
> too.
>
>
>
>
>
>
>
> > playback executes on line 87 of your trace.
>
>
>
> yes, but executed by log after 30 seconds after line 84 and  
> immediately hangup.
>
> Note, duration wav file - 4 sec.
>
>
>
> > If you do not hear the audio, It means the re-establishment of  
> media fails somehow based on your topology or
>
> the phone you are on does not support early media,
>
>
>
> I don't hear the audio, because FS don't send RTP traffic to my  
> phone in early media mode (checked sniffers) in my case.
>
> If dialplans is:
>
> <extension name="fail_balance">
>
> <condition field="${blabla}" expression="^\-1$">
>
> <action application="pre_answer"/>
>
> <action application="sleep" data="1000"/>
>
> <action application="playback" data="elight/neg_balance.wav"/>
>
> <action application="hangup" data="BEARERCAPABILITY_NOTAUTH"/>
>
> </condition>
>
> </extension>
>
> then work's fine, no problems (bypass_media=true not set).
>
>
>
> > change your pre-answer before the playback to answer to verify.
>
>
>
> If set answer, problem disappears. But answer it's not for my case.
>
>
>
> > Try adding sofia profile internal siptrace on to see the sip  
> traffic too.
>
>
>
> Call log with sip trace http://pastebin.freeswitch.org/13065
>
>
>
> Thanks again :)
>
>
>
>
>
> -- 
>
> Regard's
>
> Sergey Scheglov
>
>
>
> ---------- Forwarded message ----------
> From: Anthony Minessale <anthony.minessale at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Thu, 27 May 2010 13:46:05 -0500
> Subject: Re: [Freeswitch-users] Don't work playback after bypass  
> media mode.
> it's against the SIP spec to re-negotiate media before you have  
> answered.
> the transaction to negotiate the SDP from the original call has not  
> completed so it's illegal to send a re-invite.
>
> instead you should use bypass_media_after_bridge=true so the bypass  
> only happens when the bridge works.
>
>
> 2010/5/27 Sergey Scheglov <sid at eltc.ru>
> Hi, Anthony.
>
>
>
> You wrote 27 may 2010 ?., 20:34:15:
>
>
>
>
> playback executes on line 87 of your trace.
>
>
>
> If you do not hear the audio, It means the re-establishment of media  
> fails somehow based on your topology or
>
> the phone you are on does not support early media, change your pre- 
> answer before the playback to answer to verify.
>
>
>
> Try adding sofia profile internal siptrace on to see the sip traffic  
> too.
>
>
>
>
>
>
>
> > playback executes on line 87 of your trace.
>
>
>
> yes, but executed by log after 30 seconds after line 84 and  
> immediately hangup.
>
> Note, duration wav file - 4 sec.
>
>
>
> > If you do not hear the audio, It means the re-establishment of  
> media fails somehow based on your topology or
>
> the phone you are on does not support early media,
>
>
>
> I don't hear the audio, because FS don't send RTP traffic to my  
> phone in early media mode (checked sniffers) in my case.
>
> If dialplans is:
>
> <extension name="fail_balance">
>
> <condition field="${blabla}" expression="^\-1$">
>
> <action application="pre_answer"/>
>
> <action application="sleep" data="1000"/>
>
> <action application="playback" data="elight/neg_balance.wav"/>
>
> <action application="hangup" data="BEARERCAPABILITY_NOTAUTH"/>
>
> </condition>
>
> </extension>
>
> then work's fine, no problems (bypass_media=true not set).
>
>
>
> > change your pre-answer before the playback to answer to verify.
>
>
>
> If set answer, problem disappears. But answer it's not for my case.
>
>
>
> > Try adding sofia profile internal siptrace on to see the sip  
> traffic too.
>
>
>
> Call log with sip trace http://pastebin.freeswitch.org/13065
>
>
>
> Thanks again :)
>
>
>
>
>
> -- 
>
> Regard's
>
> Sergey Scheglov
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
>
> ---------- Forwarded message ----------
> From: Sergey Scheglov <sid at eltc.ru>
> To: Anthony Minessale <freeswitch-users at lists.freeswitch.org>
> Date: Fri, 28 May 2010 01:49:04 +0700
> Subject: Re: [Freeswitch-users] Don't work playback after bypass  
> media mode.
> Hi, Anthony.
>
> Thank's for reply.
>
> You wrote 27 may 2010 ?., 20:34:15:
>
> > playback executes on line 87 of your trace.
>
> yes, but executed by log after 30 seconds after line 84 and  
> immediately hangup.
> Note, duration wav file - 4 sec.
>
> > If you do not hear the audio, It means the re-establishment of  
> media fails
> > somehow based on your topology or
> > the phone you are on does not support early media, change your pre- 
> answer
> > before the playback to answer to verify.
>
> I don't hear the audio, because FS don't send RTP traffic to my  
> phone in early media mode (checked sniffers) in my case.
> If dialplans is:
> <extension name="fail_balance">
> <condition field="${blabla}" expression="^\-1$">
> <action application="pre_answer"/>
> <action application="sleep" data="1000"/>
> <action application="playback" data="elight/neg_balance.wav"/>
> <action application="hangup" data="BEARERCAPABILITY_NOTAUTH"/>
> </condition>
> </extension>
> then work's fine, no problems (bypass_media=true not set).
>
> If set answer, problem disappears. But answer it's not for my case.
>
> > Try adding sofia profile internal siptrace on to see the sip  
> traffic too.
>
> Call log with sip trace http://pastebin.freeswitch.org/13065
>
> Thanks again :)
>
> Best regard's
> Sergey Scheglov
>
>
>
>
>
>
> ---------- Forwarded message ----------
> From: Sergey Scheglov <sid at eltc.ru>
> To: Anthony Minessale <freeswitch-users at lists.freeswitch.org>
> Date: Fri, 28 May 2010 02:12:28 +0700
> Subject: Re: [Freeswitch-users] Don't work playback after bypass  
> media mode.
> Hi, Anthony.
>
> Thanks, thanks, thanks ))
> You wrote 28 may 2010., 1:46:05:
>
> > it's against the SIP spec to re-negotiate media before you have  
> answered.
> > the transaction to negotiate the SDP from the original call has not
> > completed so it's illegal to send a re-invite.
>
> Do I understand correctly that the dialplan is not correct?
>
> > instead you should use bypass_media_after_bridge=true so the  
> bypass only
> > happens when the bridge works.
>
> As I wrote in first message:
> "If set bypass_media_after_bridge instead bypass_media, then works
> fine, BUT changing codec negotiation." And two phones that support
> codecs, which not present in internal profile, not be able to call
> each other.
>
> Best regards
> Sergey Scheglov
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From david.ponzone at gmail.com  Fri May 28 04:31:36 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Fri, 28 May 2010 13:31:36 +0200
Subject: [Freeswitch-users] FS as Media Gateway Only
In-Reply-To: <AANLkTikgol1TigK9G9vvk90ogtjqeJ3nwPXqdFUXcV5W@mail.gmail.com>
References: <AANLkTimgKRtL06xmyjXUPnpNIjlYkHC6BytzmTqKmVm7@mail.gmail.com>
	<AANLkTikoxsXFuYBxsXXtB26Y9_8f1I1wGmE-u-I_l9PP@mail.gmail.com>
	<AANLkTikgol1TigK9G9vvk90ogtjqeJ3nwPXqdFUXcV5W@mail.gmail.com>
Message-ID: <BE75B7EE-B77D-4710-899E-AE9E771A28F7@gmail.com>

Code,

you're totally right.
In this model (FS), the media server will also be in the SIP Path.
That's why I answered in the first place that this was not achievable  
with FS, because your idea was more a Kamaillo/RTPProxy setup, where  
the mediaserver only does RTP with the endpoints, and is not in the  
SIP path at all:

inbound <--------SIP------ SIP Server/Proxy ------------SIP------->  
Carrier
                                                   |
              <---------RTP------ MediaServer--------RTP--------------->


Verizon Business (in Europe at least) has a such infrastrucutre:  
OpenSER for the SIP part, and Nortel GWs for the RTP.
This way, they just give me the IPs of their OpenSER servers, and they  
can deploy as many media servers as they need without telling us (of  
course, we dont filter that).

I don't know how this is implemented in Kamaillo/OpenSER but perhaps,  
for a nice bounty, that would be something possible in FS.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 28/05/2010 ? 05:34, Code Ghar a ?crit :

> Hi Vitalie
>
> Thanks for providing the link and details. If I understood  
> correctly, the chain of signaling would be Inbound -> FSSIP -> FSRTP  
> -> Outbound (using names and terms in my original question), while  
> the chain of media would be Inbound -> FSRTP -> Outbound. This way  
> we can have multiple servers handling media and minimal servers  
> handling signaling.
>
> Let me clarify a little more my motivation for asking this question  
> in the first place. I work with telecom carriers on a daily basis  
> and have seen many different architectures. The first biggest  
> problem is how to load balance SIP traffic when you are receiving  
> calls, if one server is not enough. The second biggest problem is  
> handling all RTP, including transcoding. With this architecture, one  
> or two IPs for signaling can be handled by most carriers. So you can  
> beef up your hardware for signaling and depend less on your  
> carrier's ability to load balance traffic for you. If they can do  
> round-robin or failover for two IPs, you are golden. And then you  
> can deploy multiple nodes to handle all RTP duties, without having  
> to concern your carrier about multiple servers and IPs. But there's  
> one thing still missing. Your outbound carrier still needs to allow  
> traffic from multiple IPs because now they are dealing with FSRTP  
> instead of FSSIP.
>
> Is there such a solution possible using FS that all signaling, for  
> both inbound and outbound carriers, can be handled by a couple of  
> FSSIP nodes (depending on the amount of traffic, maybe a few more)  
> while delegating media responsibilities to multiple FSRTP nodes? In  
> this situation, SIP IP is always, say 10.10.10.1 or 10.10.10.2, but  
> the SDP may use media IPs 10.10.10.3, 10.10.10.4, 10.10.10.5, and so  
> on. Almost all carriers I have seen concern themselves only with  
> which SIP IPs they should allow and don't care how many or which  
> media IPs they have to deal with. Any ideas on minimizing signaling  
> IPs while adding more media IPs as required?
>
> I have seen re-invite being used in production where you can just  
> let your inbound and outbound handle media between them on their own  
> without it going through your network. But there are circumstances  
> where people might need to pass media through their own network,  
> chiefly to perform transcoding and also to handle other interop  
> issues. It is because of this use case, combined with the need for  
> minimizing signaling IPs, that the original question was asked.
>
>
>
>
> On Thu, May 27, 2010 at 1:15 AM, Vitalii Colosov  
> <vetali100 at gmail.com> wrote:
> Hi Code,
>
> I have working example doing exactly what you've described.
> One signalling FS bridges incoming call to a set of media servers  
> (depending on ip, but you can implement any routing logic including  
> round robin) and then transfers media stream after bridging to that  
> media server.
>
> You can achieve this on signalling FS by creating a Lua script that  
> contains the following lines:
>
> media_server="my_media_X.mydomain.com"; --to be determined by  
> routing logic
> forwarding_session = "sofia/ 
> external/"..called_number.."@"..media_server;
> session:setVariable("bypass_media_after_bridge", "true");
> session:setVariable("hangup_after_bridge", "true");
> session:execute("bridge",forwarding_session);
>
> The call will arrive to the selected media server successfully and  
> media stream will start bypassing signalling FS after bridge.
>
> You can read the following thread, it describes how you can setup  
> such configuration.
> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-March/055231.html
>
> I think it will fit your needs.
>
> Regards,
> Vitalie
>
>
> 2010/5/27 Code Ghar <codeghar at gmail.com>
> Is it possible -- and are there any case studies, practical  
> experience, etc -- on deploying FreeSWITCH (FS) in this  
> architecture: one server (FSSIP) handles SIP signaling only, and  
> multiple servers (FSRTP1, FSRTP2, ..., FSRTPn) handle all media  
> responsibilities? So when a call comes in, the SDP contains IP of,  
> say FSRTP1, as media handler. For this to work, FSSIP would request  
> FSRTPx for media resources for each new call and add its IP and port  
> in SDP. The media servers/gateways would play IVR, etc.; collect  
> DTMF and forward as appropriate to FSSIP; perform transcoding; etc.;  
> all while FSSIP only deals with signaling. This way multiple servers  
> could be deployed to handle media responsibilities and only a  
> handful would be required for signaling. In future if there's a  
> greater need for transcoding, etc. all you need to do is deploy a  
> media server and not have to add servers for signaling.
>
> This idea came to me because I have come across two proprietary  
> applications that do it this way. They have a SIP component and a  
> media component. You can run both on the same physical machine or  
> you can separate them out into multiple machines.
>
> Another way for this could be to integrate FS as a media component  
> to another application's SIP component. A mix-and-match, so to speak.
>
> On the flip side, deploy FS as a SIP server and use media  
> capabilities of some other hardware or software application. For  
> example, FS handles signaling and use dedicated hardware for media.  
> A good example of this is illustrated (somewhat) by an image on  
> Sangoma's website: http://www.sangoma.com/assets/images/content/transcoding_diagram.jpg 
> . Look at the "pooled transcoding".
>
> Is FS even designed to be this modular? If so, how can the  
> aforementioned scenario(s) be achieved?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From codecomplete at free.fr  Fri May 28 04:41:03 2010
From: codecomplete at free.fr (GillesToo)
Date: Fri, 28 May 2010 04:41:03 -0700 (PDT)
Subject: [Freeswitch-users] Installing OpenZap?
In-Reply-To: <6F3328D2-9169-4EE7-9CC5-7D649D7E4431@tlainvestments.com>
References: <1275004306588-5111050.post@n2.nabble.com>
	<6F3328D2-9169-4EE7-9CC5-7D649D7E4431@tlainvestments.com>
Message-ID: <1275046863288-5112708.post@n2.nabble.com>



Troy Anderson-2 wrote:
> oz list will only work if openzap is loaded, so that's a clue that it
> isn't for some reason. Try issuing reload mod_openzap from the cli and
> take a look at the output.  There is likely an error on load, and that
> error may lead you to what's wrong with the config.

Thanks for the tip. Here's what "reload mod_openzap" returns:



> freeswitch at internal> load mod_openzap
> -ERR [module load file routine returned an error]
> 
> freeswitch at internal> 2010-05-28 12:13:51.138275 [DEBUG] zap_config.c:56
> Configuration file is /usr/local/freeswitch/conf/modules.conf.
> 2010-05-28 12:13:51.138275 [NOTICE] zap_io.c:3057 Modules configured: 1 
> 2010-05-28 12:13:51.139297 [DEBUG] zap_config.c:56 Configuration file is
> /usr/local/freeswitch/conf/openzap.conf.
> 2010-05-28 12:13:51.139297 [DEBUG] zap_io.c:2587 found config for span
> 2010-05-28 12:13:51.139297 [NOTICE] ozmod_zt.c:1185 Using DAHDI control
> device
> 2010-05-28 12:13:51.139297 [ERR] ozmod_zt.c:1194 Cannot open control
> device /dev/dahdi/ctl: Permission denied
> 2010-05-28 12:13:51.140310 [ERR] zap_io.c:2856 Error loading
> /usr/local/freeswitch/mod/ozmod_zt.so
> 2010-05-28 12:13:51.140310 [ERR] zap_io.c:2901 Unloading
> /usr/local/freeswitch/mod/ozmod_zt.so
> 2010-05-28 12:13:51.140310 [CRIT] zap_io.c:2610 failure creating span, no
> such type 'zt'
> 2010-05-28 12:13:51.140310 [INFO] zap_io.c:2782 Configured 0 channel(s)
> 2010-05-28 12:13:51.140310 [ERR] zap_io.c:3065 No modules configured!
> 2010-05-28 12:13:51.140310 [ERR] mod_openzap.c:3331 Error loading OpenZAP
> 2010-05-28 12:13:51.140310 [CRIT] switch_loadable_module.c:882 Error
> Loading module /usr/local/freeswitch/mod/mod_openzap.so
> **Module load routine returned an error**
> 

BTW, before I investigate further... I notice that the two articles in the
wiki about OpenZap are based on Zaptel (1.4), whose development stopped two
years ago when Zaptel was replaces with Dadhi:

http://wiki.freeswitch.org/wiki/OpenZAP
http://wiki.freeswitch.org/wiki/Zaptel_Tutorial

Has someone successfully used OpenZap with Dahdi, and if yes, are there any
changes that should be made in OpenZap to accomodate this change?

Thank you.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Installing-OpenZap-tp5111050p5112708.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Prometheus001 at gmx.net  Fri May 28 05:52:39 2010
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Fri, 28 May 2010 14:52:39 +0200
Subject: [Freeswitch-users] Any Idea how to let the phone ring with a
 different ring tone
In-Reply-To: <C01CA28A-84A7-4717-B722-DE447B51ABB2@freeswitch.org>
References: <4BFE230C.6030802@gmx.net>	<AANLkTinjGN1Nk5CP4JA6PHiGjBrbSkgEXdjskgFKrjxc@mail.gmail.com>
	<C01CA28A-84A7-4717-B722-DE447B51ABB2@freeswitch.org>
Message-ID: <4BFFBC97.4000606@gmx.net>

Thanks Brian,

I tried it, it works fine.

Best rgards
Peter



Brian West schrieb:
> This is one of the cases that doesn't follow the convention its
> "alert_info" as the variable name.
>
> /b
>
> On May 27, 2010, at 8:39 AM, Anthony Minessale wrote:
>
>> you add it as a variable in your dial string
>>
>> {sip_h_Alert-Info=<whatever it needs to
>> say>}sofia/profile/foo at bar.com <mailto:foo at bar.com>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


From lakindia89 at gmail.com  Fri May 28 06:49:39 2010
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Fri, 28 May 2010 19:19:39 +0530
Subject: [Freeswitch-users] Q931_Pcap debug - Error
Message-ID: <AANLkTinHOIrkABLt3ivzUEEH3ykgrQSOGrus3OUn0rEM@mail.gmail.com>

Hi all,
        I'm trying to configure digium with freeswitch.
        I configued zaptel and then I installed freeswitch with openzap.
        Here is my openzap.conf

        [span zt PRI_1]
        trunk_type => e1
        b-channel => 1-15
        d-channel => 16
        b-channel => 17-31

        [span zt PRI_2]
        trunk_type => e1
        b-channel => 32-46
        d-channel => 47
        b-channel => 48-62

openzap.conf.xml:
        <configuration name="openzap.conf" description="OpenZAP
Configuration">
        <settings>
        <param name="debug" value="10"/>
        </settings>
        <pri_spans>
        <span id="1" name="PRI_1">
        <param name="q921loglevel" value="debug"/>
        <param name="q931loglevel" value="debug"/>
        <param name="mode" value="user"/>
        <param name="dialect" value="q931"/>
        <param name="dialplan" value="XML"/>
        <param name="context" value="default"/>
        </span>
        <span id="2" name="PRI_2">
        <param name="q921loglevel" value="debug"/>
        <param name="q931loglevel" value="debug"/>
        <param name="mode" value="net"/>
        <param name="dialect" value="q931"/>
        <param name="dialplan" value="XML"/>
        <param name="context" value="outgoing"/>
        </span>
        </pri_spans>
        </configuration>

        I started freeswitch.
        When I said, "oz q931_pcap 1 on" in fs_cli, it tells the following
error

        2010-05-28 17:14:08.122324 [WARNING] mod_openzap.c:3070 Error
couldn't (re-)enable Q931-To-Pcap!

        I've installed libpcap and its devel package

        Can any one please say how to solve this issue?
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From tomb at cachecomm.com  Fri May 28 07:18:29 2010
From: tomb at cachecomm.com (Tom Baldwin)
Date: Fri, 28 May 2010 08:18:29 -0600
Subject: [Freeswitch-users] Installing OpenZap?
In-Reply-To: <1275046863288-5112708.post@n2.nabble.com>
References: <1275004306588-5111050.post@n2.nabble.com>
	<6F3328D2-9169-4EE7-9CC5-7D649D7E4431@tlainvestments.com>
	<1275046863288-5112708.post@n2.nabble.com>
Message-ID: <1275056309.2290.2.camel@tomb-desktop>

GillesToo,

What is the output of "dahi_scan"
can you post your openzap.conf.xml file
I run wanpipe , dahdi and openzap with libpri


Tom
 
On Fri, 2010-05-28 at 04:41 -0700, GillesToo wrote:
> 
> Troy Anderson-2 wrote:
> > oz list will only work if openzap is loaded, so that's a clue that it
> > isn't for some reason. Try issuing reload mod_openzap from the cli and
> > take a look at the output.  There is likely an error on load, and that
> > error may lead you to what's wrong with the config.
> 
> Thanks for the tip. Here's what "reload mod_openzap" returns:
> 
> 
> 
> > freeswitch at internal> load mod_openzap
> > -ERR [module load file routine returned an error]
> > 
> > freeswitch at internal> 2010-05-28 12:13:51.138275 [DEBUG] zap_config.c:56
> > Configuration file is /usr/local/freeswitch/conf/modules.conf.
> > 2010-05-28 12:13:51.138275 [NOTICE] zap_io.c:3057 Modules configured: 1 
> > 2010-05-28 12:13:51.139297 [DEBUG] zap_config.c:56 Configuration file is
> > /usr/local/freeswitch/conf/openzap.conf.
> > 2010-05-28 12:13:51.139297 [DEBUG] zap_io.c:2587 found config for span
> > 2010-05-28 12:13:51.139297 [NOTICE] ozmod_zt.c:1185 Using DAHDI control
> > device
> > 2010-05-28 12:13:51.139297 [ERR] ozmod_zt.c:1194 Cannot open control
> > device /dev/dahdi/ctl: Permission denied
> > 2010-05-28 12:13:51.140310 [ERR] zap_io.c:2856 Error loading
> > /usr/local/freeswitch/mod/ozmod_zt.so
> > 2010-05-28 12:13:51.140310 [ERR] zap_io.c:2901 Unloading
> > /usr/local/freeswitch/mod/ozmod_zt.so
> > 2010-05-28 12:13:51.140310 [CRIT] zap_io.c:2610 failure creating span, no
> > such type 'zt'
> > 2010-05-28 12:13:51.140310 [INFO] zap_io.c:2782 Configured 0 channel(s)
> > 2010-05-28 12:13:51.140310 [ERR] zap_io.c:3065 No modules configured!
> > 2010-05-28 12:13:51.140310 [ERR] mod_openzap.c:3331 Error loading OpenZAP
> > 2010-05-28 12:13:51.140310 [CRIT] switch_loadable_module.c:882 Error
> > Loading module /usr/local/freeswitch/mod/mod_openzap.so
> > **Module load routine returned an error**
> > 
> 
> BTW, before I investigate further... I notice that the two articles in the
> wiki about OpenZap are based on Zaptel (1.4), whose development stopped two
> years ago when Zaptel was replaces with Dadhi:
> 
> http://wiki.freeswitch.org/wiki/OpenZAP
> http://wiki.freeswitch.org/wiki/Zaptel_Tutorial
> 
> Has someone successfully used OpenZap with Dahdi, and if yes, are there any
> changes that should be made in OpenZap to accomodate this change?
> 
> Thank you.




From frank at impactfax.com  Fri May 28 07:27:17 2010
From: frank at impactfax.com (Frank @ Impact)
Date: Fri, 28 May 2010 10:27:17 -0400
Subject: [Freeswitch-users] sip_cid_type settings
In-Reply-To: <AANLkTikjMLzb92uv4CdOz2O977pbzmGM2IsRFE3kp05a@mail.gmail.com>
Message-ID: <C3E52E5D21AE4F5E8FC11617F733D89A@ws4>

I tried putting 

<param name="caller-id-type" value="pid"/>

in the profile settings section and also in the profile gateway section.
And in each case, FS still use rpid and seemed to ignore the param
setting.

I tested this on a fresh git from today.  Am I putting it in the wrong
place?


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: Thursday, May 27, 2010 9:45 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] sip_cid_type settings

ya, wrong param,
This is the right one:

<!-- Remote-Party-ID header -->
<param name="caller-id-type" value="rpid"/>
or

<!-- P-*-Identity family of headers -->
<param name="caller-id-type" value="pid"/>

or

<param name="caller-id-type" value="none"/>






From tomb at cachecomm.com  Fri May 28 07:32:49 2010
From: tomb at cachecomm.com (Tom Baldwin)
Date: Fri, 28 May 2010 08:32:49 -0600
Subject: [Freeswitch-users] Installing OpenZap?
In-Reply-To: <1275004306588-5111050.post@n2.nabble.com>
References: <1275004306588-5111050.post@n2.nabble.com>
Message-ID: <1275057169.2290.5.camel@tomb-desktop>

Gilles,

Sorry for the double post
In your openzap.conf
you need to have the span number like this

[span zt FXO]
name => OpenZAP
fxo-channel => 29

after the span type

Tom

On Thu, 2010-05-27 at 16:51 -0700, GillesToo wrote:
> Hello
> 
> I read the documents in the wiki, but my Dahdi + OpenZap + FXO card combo
> doesn't work, so I'd like to check if I made a mistake, overlooked
> something, or maybe some things have changed since Asterisk moved from
> Zaptel to Dahdi and the wiki hasn't been updated accordingly:
> 
> 1. Compiled Dahdi 2.3.0, and successfully loaded it:
> 
> # /etc/init.d/dahdi status
> ### Span  1: WCFXO/0 "Wildcard X100P Board 1" (MASTER) RED
>   1 FXO        FXSKS       (SWEC: MG2)  RED
>   
> 2. vi /usr/local/freeswitch/conf/zt.conf: Nothing to do here, apparently
> 
> 3. vi /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml:
> Uncommented <load module ="mod_openzap"/> 
> 
> 4. vi /usr/local/freeswitch/conf/autoload_configs/openzap.conf.xml: Left
> default settings
> 
> 5. vi /usr/local/freeswitch/conf/openzap.conf:
> [span zt]
> name => OpenZAP
> number => 1
> fxo-channel => 1
> 
> 6. The wiki says to then run fs_cli, followed by "oz list", but: "oz list:
> Command not found!"
> 
> Did I miss something?
> 
> Thank you for any hint.




From jingwei.yang at gmail.com  Fri May 28 08:05:37 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Fri, 28 May 2010 23:05:37 +0800
Subject: [Freeswitch-users] FSComm linker issue
In-Reply-To: <AANLkTimnGnnxiWW2DmMTujlOHalpjdHhOOZTMxx_DAB2@mail.gmail.com>
References: <AANLkTiliaONCmhMhVsgiXi05v5dIwASq7Zb2t-QJf1YX@mail.gmail.com>
	<AANLkTimnGnnxiWW2DmMTujlOHalpjdHhOOZTMxx_DAB2@mail.gmail.com>
Message-ID: <AANLkTik0SYCbRoOdMfYAW9seeUwcWxiDXqoFZ0ZgwhFY@mail.gmail.com>

Hi Jo?o,

Thanks a lot for the reply. Here's the jira I created:
http://jira.freeswitch.org/browse/FSCOMM-8.

Frankly speaking, I'm a total newbie to both c++ and VS2008 (just had it
installed several days back). I might not have the sufficient knowledge to
contribute. However, if you need any help, please feel free to let me know.
I'll see what I can do.

Regards,
-Jingwei

2010/5/28 Jo?o Mesquita <jmesquita at freeswitch.org>

> I must have missed that other email, I am sorry about that. The problem is
> that the VS Proj was not update with the latest code modifications I've
> made. I am the sole developer of FSComm and I develop/test in Mac/Linux.
> It's quite hard for me to keep up with all platforms by myself. Do you feel
> you could contribute with your MS knowledge? If you feel you can, I can
> provide you the pointers of what files were added and let you know.
>
> Otherwise, could you open a Jira so I won't forget to get to that when I
> have the time?
>
> Thank you for your interest on the project.
>
> Regards,
> Jo?o Mesquita
>
>
> On Thu, May 27, 2010 at 7:35 AM, Jingwei Yang <jingwei.yang at gmail.com>wrote:
>
>> Hello,
>>
>> I encountered six linker errors when compiling the latest codes of FSComm
>> in VS2008. I followed the steps from here:
>> http://wiki.freeswitch.org/wiki/FSComm#Windows. Someone else seemed to
>> have the same issue but got no answers:
>> http://article.gmane.org/gmane.comp.telephony.freeswitch.user/21757/match=fscomm
>> .
>>
>> Please enlighten me how to deal with these errors.
>>
>> Thanks and best regards,
>> -Jingwei
>>
>> 10>fshost.obj : error LNK2019: unresolved external symbol "public:
>> __thiscall Channel::Channel(class QString)" (??0Channel@@QAE at VQString@@@Z)
>> referenced in function "private: void __thiscall
>> FSHost::eventChannelCreate(class QSharedPointer<struct switch_event>,class
>> QString)" (?eventChannelCreate at FSHost
>> @@AAEXV?$QSharedPointer at Uswitch_event@@@@VQString@@@Z)
>>
>> 10>mainwindow.obj : error LNK2019: unresolved external symbol "public:
>> __thiscall StateDebugDialog::StateDebugDialog(class QWidget *)"
>> (??0StateDebugDialog@@QAE at PAVQWidget@@@Z) referenced in function
>> "private: void __thiscall MainWindow::debugEventsTriggered(void)"
>> (?debugEventsTriggered at MainWindow@@AAEXXZ)
>>
>> 10>mainwindow.obj : error LNK2019: unresolved external symbol "public:
>> __thiscall ConsoleWindow::ConsoleWindow(class QWidget *)" (??0ConsoleWindow@
>> @QAE at PAVQWidget@@@Z) referenced in function "private: void __thiscall
>> MainWindow::debugConsoleTriggered(void)" (?debugConsoleTriggered at MainWindow
>> @@AAEXXZ)
>>
>> 10>prefdialog.obj : error LNK2019: unresolved external symbol "public:
>> __thiscall CodecWidget::CodecWidget(class QWidget *)" (??0CodecWidget@
>> @QAE at PAVQWidget@@@Z) referenced in function "public: void __thiscall
>> Ui_PrefDialog::setupUi(class QDialog *)" (?setupUi at Ui_PrefDialog
>> @@QAEXPAVQDialog@@@Z)
>>
>> 10>prefsofia.obj : error LNK2019: unresolved external symbol "public: void
>> __thiscall CodecWidget::setCodecString(class QString)"
>> (?setCodecString at CodecWidget@@QAEXVQString@@@Z) referenced in function
>> "public: void __thiscall PrefSofia::readConfig(void)" (?readConfig at PrefSofia
>> @@QAEXXZ)
>>
>> 10>prefsofia.obj : error LNK2019: unresolved external symbol "public:
>> class QString __thiscall CodecWidget::getCodecString(void)"
>> (?getCodecString at CodecWidget@@QAE?AVQString@@XZ) referenced in function
>> "public: void __thiscall PrefSofia::writeConfig(void)"
>> (?writeConfig at PrefSofia@@QAEXXZ)
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Fri May 28 08:09:30 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 28 May 2010 10:09:30 -0500
Subject: [Freeswitch-users] sip_cid_type settings
In-Reply-To: <C3E52E5D21AE4F5E8FC11617F733D89A@ws4>
References: <AANLkTikjMLzb92uv4CdOz2O977pbzmGM2IsRFE3kp05a@mail.gmail.com>
	<C3E52E5D21AE4F5E8FC11617F733D89A@ws4>
Message-ID: <AANLkTik3Fl3IRqs9Y-uw29f7I13D-vxrzNL9G3hd1zHm@mail.gmail.com>

it will prefer the type used on the inbound leg.
unless you override it with the var.


On Fri, May 28, 2010 at 9:27 AM, Frank @ Impact <frank at impactfax.com> wrote:

> I tried putting
>
> <param name="caller-id-type" value="pid"/>
>
> in the profile settings section and also in the profile gateway section.
> And in each case, FS still use rpid and seemed to ignore the param
> setting.
>
> I tested this on a fresh git from today.  Am I putting it in the wrong
> place?
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Anthony Minessale
> Sent: Thursday, May 27, 2010 9:45 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] sip_cid_type settings
>
> ya, wrong param,
> This is the right one:
>
> <!-- Remote-Party-ID header -->
> <param name="caller-id-type" value="rpid"/>
> or
>
> <!-- P-*-Identity family of headers -->
> <param name="caller-id-type" value="pid"/>
>
> or
>
> <param name="caller-id-type" value="none"/>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <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
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From jcasale at activenetwerx.com  Fri May 28 08:18:37 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Fri, 28 May 2010 15:18:37 +0000
Subject: [Freeswitch-users] Installing OpenZap?
In-Reply-To: <1275046863288-5112708.post@n2.nabble.com>
References: <1275004306588-5111050.post@n2.nabble.com>
	<6F3328D2-9169-4EE7-9CC5-7D649D7E4431@tlainvestments.com>
	<1275046863288-5112708.post@n2.nabble.com>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E9A22BD2@prato.activenetwerx.local>

> **Module load routine returned an error**

Do you use udev rules? Who owns the dahdi interface, and who
does FS run as?


From codecomplete at free.fr  Fri May 28 08:22:05 2010
From: codecomplete at free.fr (GillesToo)
Date: Fri, 28 May 2010 08:22:05 -0700 (PDT)
Subject: [Freeswitch-users] Installing OpenZap?
In-Reply-To: <1275057169.2290.5.camel@tomb-desktop>
References: <1275004306588-5111050.post@n2.nabble.com>
	<1275057169.2290.5.camel@tomb-desktop>
Message-ID: <1275060125406-5113544.post@n2.nabble.com>


Thanks Tom for the help, but I'm still getting an error when typing "reload
mod_openzap" in the Freeswitch console (Actually, two errors:
"ozmod_zt.c:Cannot open control device /dev/dahdi/ctl: Permission denied",
followed by "zap_io.c failure creating span, no such type 'zt'" which could
be a syntax error of mine in openzap.conf and/or openzap.conf.xml).

Here are the outputs:

============================================
# dahdi_scan 
[1]
active=yes
alarms=OK
description=Wildcard TDM400P REV E/F Board 5
name=WCTDM/4
manufacturer=Digium
devicetype=Wildcard TDM400P REV E/F
location=PCI Bus 03 Slot 01
basechan=1
totchans=4
irq=20
type=analog
port=1,FXO
port=2,none
port=3,none
port=4,none

============================================
# /etc/init.d/dahdi status
### Span  1: WCTDM/4 "Wildcard TDM400P REV E/F Board 5" (MASTER) 
  1 FXO        FXSKS       (SWEC: MG2)  RED
  2 EMPTY                   
  3 EMPTY                   
  4 EMPTY                   

============================================
# cat /usr/local/freeswitch/conf/zt.conf 
[defaults]
codec_ms => 20
wink_ms => 150
flash_ms => 750
echo_cancel_level => 64
rxgain => 0.0
txgain => 0.0

============================================
# cat /usr/local/freeswitch/conf/openzap.conf
[span zt FXO]
name => OpenZAP
#number => 1
fxo-channel => 1

============================================
# cat /usr/local/freeswitch/conf/autoload_configs/openzap.conf.xml 
<configuration name="openzap.conf" description="OpenZAP Configuration">
  <settings>
    
    <!---->
    <!---->
    <!---->
  </settings>
   <pri_spans>
     
       <!-- Log Levels: none, alert, crit, err, warning, notice, info, debug
-->
       
       
       
       
       
       
     
     
       
       
       
       
       
       
     
   </pri_spans>
  <!-- one entry here per openzap span -->
  <analog_spans>
    <!--span id="1"-->
    <!--span id="zt"-->
    
      <!---->
      <!---->
      <!---->
      
      
      
      
      
      
      <!-- regex to stop dialing when it matches -->
      <!---->
      <!-- regex to stop dialing when it does not match -->
      <!---->
    
  </analog_spans>
</configuration>

============================================
# cat /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml 
<configuration name="modules.conf" description="Modules">
  <modules>
		[...]
    <load module="mod_openzap"/>
		[...]

============================================
#fs_cli    
freeswitch at internal> reload mod_openzap
-ERR unloading module [No such module!]
-ERR loading module [module load file routine returned an error]

freeswitch at internal> 2010-05-28 17:13:17.830127 [DEBUG] zap_config.c:56
Configuration file is /usr/local/freeswitch/conf/modules.conf.
2010-05-28 17:13:17.830127 [NOTICE] zap_io.c:3057 Modules configured: 1 
2010-05-28 17:13:17.830127 [DEBUG] zap_config.c:56 Configuration file is
/usr/local/freeswitch/conf/openzap.conf.
2010-05-28 17:13:17.830127 [DEBUG] zap_io.c:2587 found config for span
2010-05-28 17:13:17.831145 [NOTICE] ozmod_zt.c:1185 Using DAHDI control
device
2010-05-28 17:13:17.831145 [ERR] ozmod_zt.c:1194 Cannot open control device
/dev/dahdi/ctl: Permission denied
2010-05-28 17:13:17.831145 [ERR] zap_io.c:2856 Error loading
/usr/local/freeswitch/mod/ozmod_zt.so
2010-05-28 17:13:17.831145 [ERR] zap_io.c:2901 Unloading
/usr/local/freeswitch/mod/ozmod_zt.so
2010-05-28 17:13:17.831145 [CRIT] zap_io.c:2610 failure creating span, no
such type 'zt'
2010-05-28 17:13:17.831145 [INFO] zap_io.c:2782 Configured 0 channel(s)
2010-05-28 17:13:17.831145 [ERR] zap_io.c:3065 No modules configured!
2010-05-28 17:13:17.831145 [ERR] mod_openzap.c:3331 Error loading OpenZAP
2010-05-28 17:13:17.831145 [CRIT] switch_loadable_module.c:882 Error Loading
module /usr/local/freeswitch/mod/mod_openzap.so
**Module load routine returned an error**
============================================

Has someone already seen this "/dev/dahdi/ctl: Permission denied" error?

Thank you.
-- 
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Installing-OpenZap-tp5111050p5113544.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From frank at impactfax.com  Fri May 28 08:29:56 2010
From: frank at impactfax.com (Frank @ Impact)
Date: Fri, 28 May 2010 11:29:56 -0400
Subject: [Freeswitch-users] sip_cid_type settings
In-Reply-To: <AANLkTik3Fl3IRqs9Y-uw29f7I13D-vxrzNL9G3hd1zHm@mail.gmail.com>
Message-ID: <92AD23357AC440E6A370C83CF689F609@ws4>

Is the only way to override it by setting the var for the bridge?
"sip_cid_type=pid" ?
 
is there no way to override the inbound leg with the profile/gateway
settings?
 
-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: Friday, May 28, 2010 11:10 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] sip_cid_type settings
 
it will prefer the type used on the inbound leg.
unless you override it with the var.
 
On Fri, May 28, 2010 at 9:27 AM, Frank @ Impact <frank at impactfax.com>
wrote:
I tried putting

<param name="caller-id-type" value="pid"/>
in the profile settings section and also in the profile gateway section.
And in each case, FS still use rpid and seemed to ignore the param
setting.

I tested this on a fresh git from today.  Am I putting it in the wrong
place?


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: Thursday, May 27, 2010 9:45 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] sip_cid_type settings
ya, wrong param,
This is the right one:

<!-- Remote-Party-ID header -->
<param name="caller-id-type" value="rpid"/>
or

<!-- P-*-Identity family of headers -->
<param name="caller-id-type" value="pid"/>

or

<param name="caller-id-type" value="none"/>





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



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900
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From codecomplete at free.fr  Fri May 28 08:37:57 2010
From: codecomplete at free.fr (GillesToo)
Date: Fri, 28 May 2010 08:37:57 -0700 (PDT)
Subject: [Freeswitch-users] Installing OpenZap?
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E9A22BD2@prato.activenetwerx.local>
References: <1275004306588-5111050.post@n2.nabble.com>
	<6F3328D2-9169-4EE7-9CC5-7D649D7E4431@tlainvestments.com>
	<1275046863288-5112708.post@n2.nabble.com>
	<CA5A491E9DEFBE4CB777DE97E21575E9A22BD2@prato.activenetwerx.local>
Message-ID: <7.0.1.0.2.20100528173559.05e51548@free.fr>


At 17:28 28/05/2010, you wrote:
> > **Module load routine returned an error**
>Do you use udev rules? Who owns the dahdi interface, and who
>does FS run as?

Thanks Joseph for the tip. Indeed, I chowned /usr/local/freeswitch so 
all files are owned by freeswitch.freeswitch, but I compiled and 
installed dahdi as root.

How do I check file permissions for dahdi? How do I check what udev rules are? 


-- 
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From codecomplete at free.fr  Fri May 28 08:41:43 2010
From: codecomplete at free.fr (GillesToo)
Date: Fri, 28 May 2010 08:41:43 -0700 (PDT)
Subject: [Freeswitch-users] Installing OpenZap?
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E9A22BD2@prato.activenetwerx.local>
References: <1275004306588-5111050.post@n2.nabble.com>
	<6F3328D2-9169-4EE7-9CC5-7D649D7E4431@tlainvestments.com>
	<1275046863288-5112708.post@n2.nabble.com>
	<CA5A491E9DEFBE4CB777DE97E21575E9A22BD2@prato.activenetwerx.local>
Message-ID: <1275061303407-5113640.post@n2.nabble.com>


More information:

[root at freeswitch conf]# ll /dev/dahdi/
total 0
crw-rw----. 1 asterisk asterisk 196,   1 May 28 17:02 1
crw-rw----. 1 asterisk asterisk 196,   2 May 28 17:02 2
crw-rw----. 1 asterisk asterisk 196,   3 May 28 17:02 3
crw-rw----. 1 asterisk asterisk 196,   4 May 28 17:02 4
crw-rw----. 1 asterisk asterisk 196, 254 May 28 17:02 channel
crw-rw----. 1 asterisk asterisk 196,   0 May 28 17:02 ctl
crw-rw----. 1 asterisk asterisk 196, 255 May 28 17:02 pseudo
crw-rw----. 1 asterisk asterisk 196, 253 May 28 17:02 timer

I create user asterisk.asterisk because Dahdi complained at some time 
since I don't have Asterisk installed. Do you know how to chown to 
freeswitch.freeswitch?


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From jcasale at activenetwerx.com  Fri May 28 08:58:52 2010
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Fri, 28 May 2010 15:58:52 +0000
Subject: [Freeswitch-users] Installing OpenZap?
In-Reply-To: <7.0.1.0.2.20100528173559.05e51548@free.fr>
References: <1275004306588-5111050.post@n2.nabble.com>
	<6F3328D2-9169-4EE7-9CC5-7D649D7E4431@tlainvestments.com>
	<1275046863288-5112708.post@n2.nabble.com>
	<CA5A491E9DEFBE4CB777DE97E21575E9A22BD2@prato.activenetwerx.local>
	<7.0.1.0.2.20100528173559.05e51548@free.fr>
Message-ID: <CA5A491E9DEFBE4CB777DE97E21575E9A22DFE@prato.activenetwerx.local>

>Thanks Joseph for the tip. Indeed, I chowned /usr/local/freeswitch so 
>all files are owned by freeswitch.freeswitch, but I compiled and 
>installed dahdi as root. 

The chown business is fine, but who does fs actually run as? Do you use
init scripts?

# ps aux |grep freeswitch

>How do I check file permissions for dahdi? How do I check what udev rules are? 

What platform are you on? If you want, use the packages from Atrpms, Axel
builds them with a udev patch and Oslec.

You need to look in /etc/udev/rules.d/ for something dahdi related:
Yours obviously set the interface to be owned by a user/group asterisk.

FS can't access them:) See your second post, note the perms for `other`?

Edit your dahdi rule to change the owner to whoever runs fs.

HTH,
jlc


From anthony.minessale at gmail.com  Fri May 28 09:16:51 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 28 May 2010 11:16:51 -0500
Subject: [Freeswitch-users] sip_cid_type settings
In-Reply-To: <92AD23357AC440E6A370C83CF689F609@ws4>
References: <AANLkTik3Fl3IRqs9Y-uw29f7I13D-vxrzNL9G3hd1zHm@mail.gmail.com>
	<92AD23357AC440E6A370C83CF689F609@ws4>
Message-ID: <AANLkTinoKM8goKkHLD5kLpygjgbZ_Fu2rx_0FsWTRqvl@mail.gmail.com>

you can set the global var so it's always set on every call

in vars.xml

On Fri, May 28, 2010 at 10:29 AM, Frank @ Impact <frank at impactfax.com>wrote:

>  Is the only way to override it by setting the var for the bridge?
>
> "sip_cid_type=pid? ?
>
>
>
> is there no way to override the inbound leg with the profile/gateway
> settings?
>
>
>
> -----Original Message-----
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Friday, May 28, 2010 11:10 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] sip_cid_type settings
>
>
>
> it will prefer the type used on the inbound leg.
>
> unless you override it with the var.
>
>
>
> On Fri, May 28, 2010 at 9:27 AM, Frank @ Impact <frank at impactfax.com>
> wrote:
>
> I tried putting
>
>
> <param name="caller-id-type" value="pid"/>
>
> in the profile settings section and also in the profile gateway section.
> And in each case, FS still use rpid and seemed to ignore the param
> setting.
>
> I tested this on a fresh git from today.  Am I putting it in the wrong
> place?
>
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Anthony Minessale
>
> Sent: Thursday, May 27, 2010 9:45 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] sip_cid_type settings
>
> ya, wrong param,
> This is the right one:
>
> <!-- Remote-Party-ID header -->
> <param name="caller-id-type" value="rpid"/>
> or
>
> <!-- P-*-Identity family of headers -->
> <param name="caller-id-type" value="pid"/>
>
> or
>
> <param name="caller-id-type" value="none"/>
>
>
>
>
>   _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <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
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From codecomplete at free.fr  Fri May 28 09:19:23 2010
From: codecomplete at free.fr (GillesToo)
Date: Fri, 28 May 2010 09:19:23 -0700 (PDT)
Subject: [Freeswitch-users] Installing OpenZap?
In-Reply-To: <CA5A491E9DEFBE4CB777DE97E21575E9A22DFE@prato.activenetwerx.local>
References: <1275004306588-5111050.post@n2.nabble.com>
	<6F3328D2-9169-4EE7-9CC5-7D649D7E4431@tlainvestments.com>
	<1275046863288-5112708.post@n2.nabble.com>
	<CA5A491E9DEFBE4CB777DE97E21575E9A22BD2@prato.activenetwerx.local>
	<7.0.1.0.2.20100528173559.05e51548@free.fr>
	<CA5A491E9DEFBE4CB777DE97E21575E9A22DFE@prato.activenetwerx.local>
Message-ID: <1275063563917-5113780.post@n2.nabble.com>


Thanks, things look much better:

==========
# vi /etc/udev/rules.d/dahdi.rules
# /etc/init.d/dahdi restart
# ll /dev/dahdi
==========

freeswitch at internal> load mod_openzap
+OK

2010-05-28 18:15:18.865942 [DEBUG] zap_config.c:56 Configuration file is
/usr/local/freeswitch/conf/modules.conf.
freeswitch at internal> 2010-05-28 18:15:18.866962 [NOTICE] zap_io.c:3057
Modules configured: 1 
2010-05-28 18:15:18.866962 [DEBUG] zap_config.c:56 Configuration file is
/usr/local/freeswitch/conf/openzap.conf.
2010-05-28 18:15:18.866962 [DEBUG] zap_io.c:2587 found config for span
2010-05-28 18:15:18.866962 [NOTICE] ozmod_zt.c:1185 Using DAHDI control
device
2010-05-28 18:15:18.866962 [INFO] zap_io.c:2858 Loading IO from
/usr/local/freeswitch/mod/ozmod_zt.so [zt]
2010-05-28 18:15:18.866962 [DEBUG] zap_config.c:56 Configuration file is
/usr/local/freeswitch/conf/zt.conf.
2010-05-28 18:15:18.866962 [INFO] ozmod_zt.c:566 Setting rxgain val to
0.000000
2010-05-28 18:15:18.866962 [INFO] ozmod_zt.c:575 Setting txgain val to
0.000000
2010-05-28 18:15:18.868055 [INFO] zap_io.c:2604 auto-loaded 'zt'
2010-05-28 18:15:18.868055 [DEBUG] zap_io.c:2625 created span 1 (FXO) of
type zt
2010-05-28 18:15:18.868055 [DEBUG] zap_io.c:2638 span 1 [name]=[OpenZAP]
2010-05-28 18:15:18.868055 [DEBUG] zap_io.c:2638 span 1 [fxo-channel]=[1]
2010-05-28 18:15:18.868055 [DEBUG] zap_io.c:2667 setting trunk type to 'FXO'
start(KEWL)
2010-05-28 18:15:18.868055 [INFO] ozmod_zt.c:385 configuring device
/dev/dahdi/channel channel 1 as OpenZAP device 1:1 fd:51
2010-05-28 18:15:18.868055 [INFO] zap_io.c:2782 Configured 1 channel(s)
2010-05-28 18:15:18.869077 [INFO] zap_io.c:2875 Loading SIG from
/usr/local/freeswitch/mod/ozmod_analog.so
2010-05-28 18:15:18.869077 [INFO] zap_io.c:2991 auto-loaded 'analog'
2010-05-28 18:15:18.869077 [DEBUG] zap_config.c:56 Configuration file is
/usr/local/freeswitch/conf/tones.conf.
2010-05-28 18:15:18.869077 [DEBUG] zap_io.c:551 added tone generation [dial]
= [v=-7;%(1000,0,350,440)]
2010-05-28 18:15:18.869077 [DEBUG] zap_io.c:549 added tone detect [dial] =
[350,440]
2010-05-28 18:15:18.869077 [DEBUG] zap_io.c:551 added tone generation [ring]
= [v=-7;%(2000,4000,440,480)]
2010-05-28 18:15:18.869077 [DEBUG] zap_io.c:549 added tone detect [ring] =
[440,480]
2010-05-28 18:15:18.869077 [DEBUG] zap_io.c:551 added tone generation [busy]
= [v=-7;%(500,500,480,620)]
2010-05-28 18:15:18.869077 [DEBUG] zap_io.c:549 added tone detect [busy] =
[480,620]
2010-05-28 18:15:18.869077 [DEBUG] zap_io.c:551 added tone generation [attn]
= [v=0;%(100,100,1400,2060,2450,2600)]
2010-05-28 18:15:18.869077 [DEBUG] zap_io.c:549 added tone detect [attn] =
[1400,2060,2450,2600]
2010-05-28 18:15:18.869077 [DEBUG] zap_io.c:551 added tone generation
[callwaiting-sas] = [v=0;%(300,0,440)]
2010-05-28 18:15:18.869077 [DEBUG] zap_io.c:549 added tone detect
[callwaiting-sas] = [440]
2010-05-28 18:15:18.869077 [DEBUG] zap_io.c:551 added tone generation
[callwaiting-cas] = [v=0;%(80,0,2750,2130)]
2010-05-28 18:15:18.869077 [DEBUG] zap_io.c:549 added tone detect
[callwaiting-cas] = [2750,2130]
2010-05-28 18:15:18.869077 [DEBUG] zap_io.c:549 added tone detect [fail1] =
[913.8]
2010-05-28 18:15:18.869077 [DEBUG] zap_io.c:549 added tone detect [fail2] =
[1370.6]
2010-05-28 18:15:18.869077 [DEBUG] zap_io.c:549 added tone detect [fail3] =
[1776.7]
2010-05-28 18:15:18.870091 [ERR] mod_openzap.c:2410 Error finding OpenZAP
span id: name:PRI_1
2010-05-28 18:15:18.870091 [ERR] mod_openzap.c:2410 Error finding OpenZAP
span id: name:PRI_2
2010-05-28 18:15:18.870091 [DEBUG] ozmod_analog.c:946 ANALOG thread
starting.
2010-05-28 18:15:18.870091 [CONSOLE] switch_loadable_module.c:900
Successfully Loaded [mod_openzap]
2010-05-28 18:15:18.870091 [NOTICE] switch_loadable_module.c:144 Adding
Endpoint 'openzap'
2010-05-28 18:15:18.870091 [NOTICE] switch_loadable_module.c:250 Adding
Application 'disable_ec'
2010-05-28 18:15:18.870091 [NOTICE] switch_loadable_module.c:250 Adding
Application 'disable_dtmf'
2010-05-28 18:15:18.870091 [NOTICE] switch_loadable_module.c:250 Adding
Application 'enable_dtmf'
2010-05-28 18:15:18.870091 [NOTICE] switch_loadable_module.c:272 Adding API
Function 'oz'
==========

Since I don't have a digital interface, I guess I can remove the PRI_1 and
PRI_2 spans from 
conf/autoload_configs/openzap.conf.xml to avoid the two ERR above.

On to configuring Freeswitch to handle incoming calls from the FXO port...

Thanks everyone.
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From frank at impactfax.com  Fri May 28 10:43:55 2010
From: frank at impactfax.com (Frank @ Impact)
Date: Fri, 28 May 2010 13:43:55 -0400
Subject: [Freeswitch-users] how to get session of bridged Bleg
Message-ID: <7DBF18332A1F49869C7B797676DFE258@ws4>

FreeSWITCH Version 1.0.head (git-) from 5/28/10
 
I have a lua script called from the dialplan that bridges the call with 
session:execute("bridge","{route=sofia/gateway/US/15555551212}
sofia/gateway/US/15555551212")
 
after the call ends, I am trying to get some variables out of the Bleg
session, the bridged leg.  But I do not have a session handle to get the
variables.  I know what the uuid is from global variable_bridge_uuid.
But how can I get the session of the B-leg so that I can do a
getVariable()?  The variables I want are not available in any of the
global variables I can see with info.
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From anthony.minessale at gmail.com  Fri May 28 12:41:29 2010
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 28 May 2010 14:41:29 -0500
Subject: [Freeswitch-users] how to get session of bridged Bleg
In-Reply-To: <7DBF18332A1F49869C7B797676DFE258@ws4>
References: <7DBF18332A1F49869C7B797676DFE258@ws4>
Message-ID: <AANLkTinj_5ooO4a8OW2LDn4hEHRGOi5FFeqVBxMFdz9X@mail.gmail.com>

you can't, at least the way you are trying.

you should not be doing accounting in your script, rather, you should make a
separate system to process
call data.




On Fri, May 28, 2010 at 12:43 PM, Frank @ Impact <frank at impactfax.com>wrote:

>  FreeSWITCH Version 1.0.head (git-) from 5/28/10
>
>
>
> I have a lua script called from the dialplan that bridges the call with
>
> session:execute("bridge","{route=sofia/gateway/US/15555551212}
> sofia/gateway/US/15555551212")
>
>
>
> after the call ends, I am trying to get some variables out of the Blegsession, the bridged leg.
>  But I do not have a session handle to get the variables.  I know what the
> uuid is from global variable_bridge_uuid.  But how can I get the session
> of the B-leg so that I can do a getVariable()?  The variables I want are
> not available in any of the global variables I can see with info.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com <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
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From frank at impactfax.com  Fri May 28 13:29:58 2010
From: frank at impactfax.com (Frank @ Impact)
Date: Fri, 28 May 2010 16:29:58 -0400
Subject: [Freeswitch-users] how to get session of bridged Bleg
In-Reply-To: <AANLkTinj_5ooO4a8OW2LDn4hEHRGOi5FFeqVBxMFdz9X@mail.gmail.com>
Message-ID: <CB6C2EE3BDAE4AA5B08A8F1D551D3424@ws4>

I am not trying to do accounting in the lua script.  
I am really just trying to determine which leg hung up on the bridge.
Is there another way to get that same information if it do it like I am
trying?
 
-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: Friday, May 28, 2010 3:41 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] how to get session of bridged Bleg
 
you can't, at least the way you are trying.
 
you should not be doing accounting in your script, rather, you should
make a separate system to process
call data.
 
 
 
On Fri, May 28, 2010 at 12:43 PM, Frank @ Impact <frank at impactfax.com>
wrote:
FreeSWITCH Version 1.0.head (git-) from 5/28/10
 
I have a lua script called from the dialplan that bridges the call with 
session:execute("bridge","{route=sofia/gateway/US/15555551212}
sofia/gateway/US/15555551212")
 
after the call ends, I am trying to get some variables out of the Bleg
session, the bridged leg.  But I do not have a session handle to get the
variables.  I know what the uuid is from global variable_bridge_uuid.
But how can I get the session of the B-leg so that I can do a
getVariable()?  The variables I want are not available in any of the
global variables I can see with info.

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



-- 
Anthony Minessale II

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

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

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:+19193869900
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From marketing at cluecon.com  Fri May 28 17:24:58 2010
From: marketing at cluecon.com (Michael Collins)
Date: Fri, 28 May 2010 17:24:58 -0700
Subject: [Freeswitch-users] Register On-Line For ClueCon!
Message-ID: <AANLkTinO73IEHIbs43V14Mgg7_THutsJ6yWvLNuQDmGD@mail.gmail.com>

Hey there! We just wanted to let everyone know that our Web site has been
updated:

http://www.cluecon.com

You may now click the link to get registered. Remember that you need to
create a user for the ClueCon Web site or use the same username/password
that you used last year. Feel free to contact us if you have any concerns or
questions.

Looking forward to Chicago!
The ClueCon Team
http://www.cluecon.com
877.742.CLUE
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From b_ball_henry at hotmail.com  Fri May 28 20:19:13 2010
From: b_ball_henry at hotmail.com (Henry Huang)
Date: Sat, 29 May 2010 11:19:13 +0800
Subject: [Freeswitch-users] how to get session of bridged Bleg
In-Reply-To: <AANLkTinj_5ooO4a8OW2LDn4hEHRGOi5FFeqVBxMFdz9X@mail.gmail.com>
References: <7DBF18332A1F49869C7B797676DFE258@ws4>
	<AANLkTinj_5ooO4a8OW2LDn4hEHRGOi5FFeqVBxMFdz9X@mail.gmail.com>
Message-ID: <AANLkTilBNpp70KiFKQuuH_lUB8U4WYDRPguioqAvyI-o@mail.gmail.com>

Anthony:

Can you kindly describe why is it not a good practice to do accounting in
the script? And what's the major benefit for doing accounting from a
separate system?

Henry

Chat Skype: unicsolution MSN: b_ball_henry at hotmail.com
Contact Me [image:
Linkedin]<http://www.linkedin.com/pub/henry-huang/1/654/578>[image:
Facebook] <http://www.facebook.com/profile.php?id=595148342>[image:
Twitter]<http://twitter.com/unicsolution>


On Sat, May 29, 2010 at 3:41 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> you can't, at least the way you are trying.
>
> you should not be doing accounting in your script, rather, you should make
> a separate system to process
> call data.
>
>
>
>
> On Fri, May 28, 2010 at 12:43 PM, Frank @ Impact <frank at impactfax.com>wrote:
>
>>  FreeSWITCH Version 1.0.head (git-) from 5/28/10
>>
>>
>>
>> I have a lua script called from the dialplan that bridges the call with
>>
>> session:execute("bridge","{route=sofia/gateway/US/15555551212}
>> sofia/gateway/US/15555551212")
>>
>>
>>
>> after the call ends, I am trying to get some variables out of the Blegsession, the bridged leg.
>>  But I do not have a session handle to get the variables.  I know what
>> the uuid is from global variable_bridge_uuid.  But how can I get the
>> session of the B-leg so that I can do a getVariable()?  The variables I
>> want are not available in any of the global variables I can see with info.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <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-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Fri May 28 20:56:40 2010
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 28 May 2010 20:56:40 -0700
Subject: [Freeswitch-users] how to get session of bridged Bleg
In-Reply-To: <AANLkTilBNpp70KiFKQuuH_lUB8U4WYDRPguioqAvyI-o@mail.gmail.com>
References: <7DBF18332A1F49869C7B797676DFE258@ws4>
	<AANLkTinj_5ooO4a8OW2LDn4hEHRGOi5FFeqVBxMFdz9X@mail.gmail.com>
	<AANLkTilBNpp70KiFKQuuH_lUB8U4WYDRPguioqAvyI-o@mail.gmail.com>
Message-ID: <AANLkTikINNFATV-bLEx1dJ23t5qNtKS4QXRob6ZuHC-K@mail.gmail.com>

On Fri, May 28, 2010 at 8:19 PM, Henry Huang <b_ball_henry at hotmail.com>wrote:

> Anthony:
>
> Can you kindly describe why is it not a good practice to do accounting in
> the script? And what's the major benefit for doing accounting from a
> separate system?
>
> "Separation of concerns." The PBX/soft-switch is designed to connect calls
and it does a great job of that. It also does a great job of producing raw
information from which a proper database application can do billing. Mixing
the two has, at best, marginal benefits, but it can cause many troubles.
Having the billing app/db in a separate process (preferably on a completely
separate system) makes it more robust, more scalable, easier to maintain and
easier to troubleshoot. Those rewards are well worth the relatively small
effort that is required to build the billing system separately.

-MC


> Henry
>
> Chat Skype: unicsolution MSN: b_ball_henry at hotmail.com
> Contact Me [image: Linkedin]<http://www.linkedin.com/pub/henry-huang/1/654/578>[image:
> Facebook] <http://www.facebook.com/profile.php?id=595148342>[image:
> Twitter] <http://twitter.com/unicsolution>
>
>
> On Sat, May 29, 2010 at 3:41 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> you can't, at least the way you are trying.
>>
>> you should not be doing accounting in your script, rather, you should make
>> a separate system to process
>> call data.
>>
>>
>>
>>
>> On Fri, May 28, 2010 at 12:43 PM, Frank @ Impact <frank at impactfax.com>wrote:
>>
>>>  FreeSWITCH Version 1.0.head (git-) from 5/28/10
>>>
>>>
>>>
>>> I have a lua script called from the dialplan that bridges the call with
>>>
>>> session:execute("bridge","{route=sofia/gateway/US/15555551212}
>>> sofia/gateway/US/15555551212")
>>>
>>>
>>>
>>> after the call ends, I am trying to get some variables out of the Blegsession, the bridged leg.
>>>  But I do not have a session handle to get the variables.  I know what
>>> the uuid is from global variable_bridge_uuid.  But how can I get the
>>> session of the B-leg so that I can do a getVariable()?  The variables I
>>> want are not available in any of the global variables I can see with info.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <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-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From babak.freeswitch at gmail.com  Fri May 28 23:41:11 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Sat, 29 May 2010 11:11:11 +0430
Subject: [Freeswitch-users] streamFile() problem
Message-ID: <AANLkTimiaq9NZTDlGyMeTl9w45UForwlJA_h3VS3plnY@mail.gmail.com>

Hi
using session.streamfile() , caller hears the playback from the sec 5 or 6
not from the start. it seems the file was playing but the media is not sent
to caller till sec 5 6
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From babak.freeswitch at gmail.com  Sat May 29 05:58:53 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Sat, 29 May 2010 17:28:53 +0430
Subject: [Freeswitch-users] nat problem!
In-Reply-To: <BEF3111D-C78F-4BF3-9C26-D066EC128A66@gmail.com>
References: <AANLkTin2PsoyY9jqaq2CW6EY-IfyyUxSUQFPnHBGaQuA@mail.gmail.com>
	<880247BE-772D-4595-B591-D0577D9F9756@freeswitch.org>
	<AANLkTinthmE7uLkV4g_zcuWYiyBQ4wiDf8zdXaUnpW5M@mail.gmail.com>
	<AANLkTimG25bRUAlc8KwB8v9XjQM7xqxbnnT5OnE7mC34@mail.gmail.com>
	<F1C6AB5C-545A-487F-B660-4168A1E16219@gmail.com>
	<AANLkTikBTfJbSaWLA_7EUMNPBYlNZbkPEKLbJF8y64Pm@mail.gmail.com>
	<AANLkTilNIDxTurbL7CGnUvMjkbdDi2kSZD2EhDaodP-q@mail.gmail.com>
	<BEF3111D-C78F-4BF3-9C26-D066EC128A66@gmail.com>
Message-ID: <AANLkTikhUnNW9RfnZxKnLzV_hUCJuNhnFnYLaF1inw0L@mail.gmail.com>

thanx for ur answers
but when I set ndlb force rport to true freeswitch responses with:
5060 (0x13C4) 49834 (0xC2AA) 192.168.11.30 192.168.7.211 SIP SIP:Response:
SIP/2.0 407 Proxy Authentication Required
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From brian at freeswitch.org  Sat May 29 06:06:44 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 29 May 2010 08:06:44 -0500
Subject: [Freeswitch-users] nat problem!
In-Reply-To: <AANLkTikhUnNW9RfnZxKnLzV_hUCJuNhnFnYLaF1inw0L@mail.gmail.com>
References: <AANLkTin2PsoyY9jqaq2CW6EY-IfyyUxSUQFPnHBGaQuA@mail.gmail.com>
	<880247BE-772D-4595-B591-D0577D9F9756@freeswitch.org>
	<AANLkTinthmE7uLkV4g_zcuWYiyBQ4wiDf8zdXaUnpW5M@mail.gmail.com>
	<AANLkTimG25bRUAlc8KwB8v9XjQM7xqxbnnT5OnE7mC34@mail.gmail.com>
	<F1C6AB5C-545A-487F-B660-4168A1E16219@gmail.com>
	<AANLkTikBTfJbSaWLA_7EUMNPBYlNZbkPEKLbJF8y64Pm@mail.gmail.com>
	<AANLkTilNIDxTurbL7CGnUvMjkbdDi2kSZD2EhDaodP-q@mail.gmail.com>
	<BEF3111D-C78F-4BF3-9C26-D066EC128A66@gmail.com>
	<AANLkTikhUnNW9RfnZxKnLzV_hUCJuNhnFnYLaF1inw0L@mail.gmail.com>
Message-ID: <D2590FAD-3115-40D3-BD0F-268F1E5A4846@freeswitch.org>

That means its send a challenge and the far end isn't answering check your username or password.

/b

On May 29, 2010, at 7:58 AM, babak yakhchali wrote:

> thanx for ur answers
> but when I set ndlb force rport to true freeswitch responses with:
> 5060 (0x13C4)	49834 (0xC2AA)	192.168.11.30	192.168.7.211	SIP	SIP:Response: SIP/2.0 407 Proxy Authentication Required
> 

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From babak.freeswitch at gmail.com  Sat May 29 06:19:38 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Sat, 29 May 2010 17:49:38 +0430
Subject: [Freeswitch-users] nat problem!
In-Reply-To: <D2590FAD-3115-40D3-BD0F-268F1E5A4846@freeswitch.org>
References: <AANLkTin2PsoyY9jqaq2CW6EY-IfyyUxSUQFPnHBGaQuA@mail.gmail.com>
	<880247BE-772D-4595-B591-D0577D9F9756@freeswitch.org>
	<AANLkTinthmE7uLkV4g_zcuWYiyBQ4wiDf8zdXaUnpW5M@mail.gmail.com>
	<AANLkTimG25bRUAlc8KwB8v9XjQM7xqxbnnT5OnE7mC34@mail.gmail.com>
	<F1C6AB5C-545A-487F-B660-4168A1E16219@gmail.com>
	<AANLkTikBTfJbSaWLA_7EUMNPBYlNZbkPEKLbJF8y64Pm@mail.gmail.com>
	<AANLkTilNIDxTurbL7CGnUvMjkbdDi2kSZD2EhDaodP-q@mail.gmail.com>
	<BEF3111D-C78F-4BF3-9C26-D066EC128A66@gmail.com>
	<AANLkTikhUnNW9RfnZxKnLzV_hUCJuNhnFnYLaF1inw0L@mail.gmail.com>
	<D2590FAD-3115-40D3-BD0F-268F1E5A4846@freeswitch.org>
Message-ID: <AANLkTimnubtYMhVvSx3seX5tipF-Vcy_2nbuFay06r_3@mail.gmail.com>

Hi brian
I'm in a really bad situation. as I said before when I captured packets I
found out:
on first invite: every things good (responses to ip phones are sent to 5060
and ipphones work fine)
on reinvites(invites during the call): problem! (invites are sent to 48796
and no response come from ipphones)
on bye: (byes are sent to 48899):problem same as above!
why freeswitch is changing behaviour??
thanx
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From frank at impactfax.com  Sat May 29 07:48:17 2010
From: frank at impactfax.com (Frank @ Impact)
Date: Sat, 29 May 2010 10:48:17 -0400
Subject: [Freeswitch-users] how to get session of bridged Bleg
In-Reply-To: <AANLkTikINNFATV-bLEx1dJ23t5qNtKS4QXRob6ZuHC-K@mail.gmail.com>
Message-ID: <A38BE7E91DB648CEB6E824194149542F@ws4>

Michael,
 
I agree to do the accounting work somewhere else.  But I am really just
trying to determine which leg hung up on the bridge when I do
session:execute("bridge","{route=sofia/gateway/US/15555551212}
sofia/gateway/US/15555551212")
I am trying to get the sip_hangup_disposition from both legs.  The
variables for both legs is not in the 'info' output available after the
bridge ends.
 
Is there another way to get this same information if I do the bridge as
listed?
 
-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: Friday, May 28, 2010 11:57 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] how to get session of bridged Bleg
 
 
On Fri, May 28, 2010 at 8:19 PM, Henry Huang <b_ball_henry at hotmail.com>
wrote:
Anthony:

Can you kindly describe why is it not a good practice to do accounting
in the script? And what's the major benefit for doing accounting from a
separate system?
"Separation of concerns." The PBX/soft-switch is designed to connect
calls and it does a great job of that. It also does a great job of
producing raw information from which a proper database application can
do billing. Mixing the two has, at best, marginal benefits, but it can
cause many troubles. Having the billing app/db in a separate process
(preferably on a completely separate system) makes it more robust, more
scalable, easier to maintain and easier to troubleshoot. Those rewards
are well worth the relatively small effort that is required to build the
billing system separately.

-MC
 
Henry
Chat Skype: unicsolution MSN: b_ball_henry at hotmail.com
Contact Me  <http://www.linkedin.com/pub/henry-huang/1/654/578> Linkedin
<http://www.facebook.com/profile.php?id=595148342> Facebook
<http://twitter.com/unicsolution> Twitter
 
On Sat, May 29, 2010 at 3:41 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
you can't, at least the way you are trying.
 
you should not be doing accounting in your script, rather, you should
make a separate system to process
call data.
 
 
 
On Fri, May 28, 2010 at 12:43 PM, Frank @ Impact <frank at impactfax.com>
wrote:
FreeSWITCH Version 1.0.head (git-) from 5/28/10
 
I have a lua script called from the dialplan that bridges the call with 
session:execute("bridge","{route=sofia/gateway/US/15555551212}
sofia/gateway/US/15555551212")
 
after the call ends, I am trying to get some variables out of the Bleg
session, the bridged leg.  But I do not have a session handle to get the
variables.  I know what the uuid is from global variable_bridge_uuid.
But how can I get the session of the B-leg so that I can do a
getVariable()?  The variables I want are not available in any of the
global variables I can see with info.
 
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



-- 
Anthony Minessale II

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

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

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

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




_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
 
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From frank at impactfax.com  Sat May 29 08:05:54 2010
From: frank at impactfax.com (Frank @ Impact)
Date: Sat, 29 May 2010 11:05:54 -0400
Subject: [Freeswitch-users] how to get session of bridged Bleg
In-Reply-To: <A38BE7E91DB648CEB6E824194149542F@ws4>
Message-ID: <6DB4E9CA948342B6A63D3DB610C3C9DC@ws4>

Or
 is there something I could do in my lua script that is setting up
the bridge if I used 
bridge_pre_execute_bleg_app to somehow get the bleg session object so
that I can refer to it later in the lua script after the bridge has
terminated?  I am not so sure how to use bridge_pre_execute_bleg_app
from within the lua to do this.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Frank @ Impact
Sent: Saturday, May 29, 2010 10:48 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] how to get session of bridged Bleg

Michael,

I agree to do the accounting work somewhere else.  But I am really just
trying to determine which leg hung up on the bridge when I do
session:execute("bridge","{route=sofia/gateway/US/15555551212}
sofia/gateway/US/15555551212")
I am trying to get the sip_hangup_disposition from both legs.  The
variables for both legs is not in the ?info? output available after the
bridge ends.

Is there another way to get this same information if I do the bridge as
listed?

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: Friday, May 28, 2010 11:57 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] how to get session of bridged Bleg


On Fri, May 28, 2010 at 8:19 PM, Henry Huang <b_ball_henry at hotmail.com>
wrote:
Anthony:

Can you kindly describe why is it not a good practice to do accounting
in the script? And what's the major benefit for doing accounting from a
separate system?
"Separation of concerns." The PBX/soft-switch is designed to connect
calls and it does a great job of that. It also does a great job of
producing raw information from which a proper database application can
do billing. Mixing the two has, at best, marginal benefits, but it can
cause many troubles. Having the billing app/db in a separate process
(preferably on a completely separate system) makes it more robust, more
scalable, easier to maintain and easier to troubleshoot. Those rewards
are well worth the relatively small effort that is required to build the
billing system separately.

-MC
?
Henry
Chat Skype: unicsolution MSN: b_ball_henry at hotmail.com
Contact Me 

On Sat, May 29, 2010 at 3:41 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
you can't, at least the way you are trying.

you should not be doing accounting in your script, rather, you should
make a separate system to process
call data.



On Fri, May 28, 2010 at 12:43 PM, Frank @ Impact <frank at impactfax.com>
wrote:
FreeSWITCH Version 1.0.head (git-) from 5/28/10
?
I have a lua script called from the dialplan that bridges the call with 
session:execute("bridge","{route=sofia/gateway/US/15555551212}
sofia/gateway/US/15555551212")
?
after the call ends, I am trying to get some variables out of the Bleg
session, the bridged leg. ?But I do not have a session handle to get the
variables.? I know what the uuid is from global variable_bridge_uuid.?
But how can I get the session of the B-leg so that I can do a
getVariable()?? The variables I want are not available in any of the
global variables I can see with info.

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



-- 
Anthony Minessale II

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

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

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

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




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




From brian at freeswitch.org  Sat May 29 08:07:06 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 29 May 2010 10:07:06 -0500
Subject: [Freeswitch-users] how to get session of bridged Bleg
In-Reply-To: <A38BE7E91DB648CEB6E824194149542F@ws4>
References: <A38BE7E91DB648CEB6E824194149542F@ws4>
Message-ID: <BFFB0190-2C07-4AA2-8DD9-A3166862287E@freeswitch.org>

You can't get the hangup disposition till the channel is hung up.  That is why they call it hangup disposition.

/b

Sent from my iPad

On May 29, 2010, at 9:48 AM, "Frank @ Impact" <frank at impactfax.com> wrote:

> Michael,
>  
> I agree to do the accounting work somewhere else.  But I am really just trying to determine which leg hung up on the bridge when I do
> session:execute("bridge","{route=sofia/gateway/US/15555551212} sofia/gateway/US/15555551212")
> I am trying to get the sip_hangup_disposition from both legs.  The variables for both legs is not in the ?info? output available after the bridge ends.
>  
> Is there another way to get this same information if I do the bridge as listed?
>  
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
> Sent: Friday, May 28, 2010 11:57 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] how to get session of bridged Bleg
>  
>  
> 
> On Fri, May 28, 2010 at 8:19 PM, Henry Huang <b_ball_henry at hotmail.com> wrote:
> Anthony:
> 
> Can you kindly describe why is it not a good practice to do accounting in the script? And what's the major benefit for doing accounting from a separate system?
> 
> "Separation of concerns." The PBX/soft-switch is designed to connect calls and it does a great job of that. It also does a great job of producing raw information from which a proper database application can do billing. Mixing the two has, at best, marginal benefits, but it can cause many troubles. Having the billing app/db in a separate process (preferably on a completely separate system) makes it more robust, more scalable, easier to maintain and easier to troubleshoot. Those rewards are well worth the relatively small effort that is required to build the billing system separately.
> 
> -MC
>  
> Henry
> 
> Chat Skype: unicsolution MSN: b_ball_henry at hotmail.com
> Contact Me 
>  
> 
> On Sat, May 29, 2010 at 3:41 AM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
> you can't, at least the way you are trying.
>  
> you should not be doing accounting in your script, rather, you should make a separate system to process
> call data.
>  
>  
>  
> 
> On Fri, May 28, 2010 at 12:43 PM, Frank @ Impact <frank at impactfax.com> wrote:
> FreeSWITCH Version 1.0.head (git-) from 5/28/10
>  
> I have a lua script called from the dialplan that bridges the call with
> session:execute("bridge","{route=sofia/gateway/US/15555551212} sofia/gateway/US/15555551212")
>  
> after the call ends, I am trying to get some variables out of the Bleg session, the bridged leg.  But I do not have a session handle to get the variables.  I know what the uuid is from global variable_bridge_uuid.  But how can I get the session of the B-leg so that I can do a getVariable()?  The variables I want are not available in any of the global variables I can see with info.
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
> 
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From frank at impactfax.com  Sat May 29 10:21:18 2010
From: frank at impactfax.com (Frank @ Impact)
Date: Sat, 29 May 2010 13:21:18 -0400
Subject: [Freeswitch-users] how to get session of bridged Bleg
In-Reply-To: <BFFB0190-2C07-4AA2-8DD9-A3166862287E@freeswitch.org>
Message-ID: <AB323555E1AC4C86899C760031FE6C33@ws4>

Yes.  I understand that.  So, after the bridge is done, how can I get
the hangup disposition of both legs?  Again, I am just trying to record
which leg said bye.
 
-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Brian West
Sent: Saturday, May 29, 2010 11:07 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] how to get session of bridged Bleg
 
You can't get the hangup disposition till the channel is hung up.  That
is why they call it hangup disposition.
/b

 
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From brian at freeswitch.org  Sat May 29 10:44:24 2010
From: brian at freeswitch.org (Brian West)
Date: Sat, 29 May 2010 12:44:24 -0500
Subject: [Freeswitch-users] how to get session of bridged Bleg
In-Reply-To: <AB323555E1AC4C86899C760031FE6C33@ws4>
References: <AB323555E1AC4C86899C760031FE6C33@ws4>
Message-ID: <38878699-4260-458F-839C-20084B5BEE6C@freeswitch.org>

you can't till the channel is hung up.

/b

Sent from my iPad

On May 29, 2010, at 12:21 PM, "Frank @ Impact" <frank at impactfax.com> wrote:

> Yes.  I understand that.  So, after the bridge is done, how can I get the hangup disposition of both legs?  Again, I am just trying to record which leg said bye.
>  
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
> Sent: Saturday, May 29, 2010 11:07 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] how to get session of bridged Bleg
>  
> You can't get the hangup disposition till the channel is hung up.  That is why they call it hangup disposition.
> 
> /b
> 
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From msc at freeswitch.org  Sat May 29 12:26:58 2010
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 29 May 2010 12:26:58 -0700
Subject: [Freeswitch-users] how to get session of bridged Bleg
In-Reply-To: <AB323555E1AC4C86899C760031FE6C33@ws4>
References: <BFFB0190-2C07-4AA2-8DD9-A3166862287E@freeswitch.org>
	<AB323555E1AC4C86899C760031FE6C33@ws4>
Message-ID: <AANLkTimrjhRhYo8H4LeJaQkgGv6sB-cWuDgAsl7knlfa@mail.gmail.com>

On Sat, May 29, 2010 at 10:21 AM, Frank @ Impact <frank at impactfax.com>wrote:

>  Yes.  I understand that.  So, after the bridge is done, how can I get the
> hangup disposition of both legs?  Again, I am just trying to record which
> leg said bye.
>
Okay, I think the point that Brian is trying to drive home is that since the
hangup disposition isn't known until hangup occurs you cannot know the
disposition inside of a session. The session ceases to exist at hangup so
the only way to know the disposition is to look at the CDRs or to play
around with "set_hangup_hook" and "session_in_hangup_hook" chan vars.
-MC

>
>
> -----Original Message-----
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West
> *Sent:* Saturday, May 29, 2010 11:07 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] how to get session of bridged Bleg
>
>
>
> You can't get the hangup disposition till the channel is hung up.  That is
> why they call it hangup disposition.
>
> /b
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Sat May 29 12:33:32 2010
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 29 May 2010 12:33:32 -0700
Subject: [Freeswitch-users] nat problem!
In-Reply-To: <AANLkTimnubtYMhVvSx3seX5tipF-Vcy_2nbuFay06r_3@mail.gmail.com>
References: <AANLkTin2PsoyY9jqaq2CW6EY-IfyyUxSUQFPnHBGaQuA@mail.gmail.com>
	<880247BE-772D-4595-B591-D0577D9F9756@freeswitch.org>
	<AANLkTinthmE7uLkV4g_zcuWYiyBQ4wiDf8zdXaUnpW5M@mail.gmail.com>
	<AANLkTimG25bRUAlc8KwB8v9XjQM7xqxbnnT5OnE7mC34@mail.gmail.com>
	<F1C6AB5C-545A-487F-B660-4168A1E16219@gmail.com>
	<AANLkTikBTfJbSaWLA_7EUMNPBYlNZbkPEKLbJF8y64Pm@mail.gmail.com>
	<AANLkTilNIDxTurbL7CGnUvMjkbdDi2kSZD2EhDaodP-q@mail.gmail.com>
	<BEF3111D-C78F-4BF3-9C26-D066EC128A66@gmail.com>
	<AANLkTikhUnNW9RfnZxKnLzV_hUCJuNhnFnYLaF1inw0L@mail.gmail.com>
	<D2590FAD-3115-40D3-BD0F-268F1E5A4846@freeswitch.org>
	<AANLkTimnubtYMhVvSx3seX5tipF-Vcy_2nbuFay06r_3@mail.gmail.com>
Message-ID: <AANLkTil6xFDgtEH10rr353o2aJUiFmM311mMgDD6JhIu@mail.gmail.com>

Insufficient information. You need to pastebin everything relevant to the
situation:
Topology - the layout of your network, with routers/firewalls as well as the
SIP endpoints
Model numbers of all devices doing NAT in this scenario
Model numbers of all IP phones in this scenario
The complete SIP trace from start to finish

-MC

On Sat, May 29, 2010 at 6:19 AM, babak yakhchali <babak.freeswitch at gmail.com
> wrote:

> Hi brian
> I'm in a really bad situation. as I said before when I captured packets I
> found out:
> on first invite: every things good (responses to ip phones are sent to 5060
> and ipphones work fine)
> on reinvites(invites during the call): problem! (invites are sent to 48796
> and no response come from ipphones)
> on bye: (byes are sent to 48899):problem same as above!
> why freeswitch is changing behaviour??
> thanx
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Sat May 29 12:35:14 2010
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 29 May 2010 12:35:14 -0700
Subject: [Freeswitch-users] streamFile() problem
In-Reply-To: <AANLkTimiaq9NZTDlGyMeTl9w45UForwlJA_h3VS3plnY@mail.gmail.com>
References: <AANLkTimiaq9NZTDlGyMeTl9w45UForwlJA_h3VS3plnY@mail.gmail.com>
Message-ID: <AANLkTikHbwPi_RDGbZJnW4lOHAfryWxabQsBICOD1D0R@mail.gmail.com>

Like your other post this is insufficient information for a diagnosis.
Please give us more details. Pastebin a console log including a SIP trace.
See this page for further instructions:
http://wiki.freeswitch.org/wiki/Reporting_Bugs

-MC

On Fri, May 28, 2010 at 11:41 PM, babak yakhchali <
babak.freeswitch at gmail.com> wrote:

> Hi
> using session.streamfile() , caller hears the playback from the sec 5 or 6
> not from the start. it seems the file was playing but the media is not sent
> to caller till sec 5 6
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From d at d-man.org  Sat May 29 12:55:33 2010
From: d at d-man.org (Darren Schreiber)
Date: Sat, 29 May 2010 12:55:33 -0700
Subject: [Freeswitch-users] 4 WEEKS LEFT! FreeSWITCH Official Training
 Course - June 28th-30th in San Francisco, CA
Message-ID: <8A034A3098ED3C4990F7D9DE40F5585F17DB0D1A64@EXVMBX020-3.exch020.serverdata.net>

Hi folks,
    In case you haven't heard, we're pleased to offer the first ever Official FreeSWITCH Training course, held June 28th-30th in San Francisco, CA. The class has been filling up (and airline tickets will go up, too) so if you are thinking of attending we *strongly* encourage you to sign-up now.

    The training is different then a basic PBX training - we'll be diving a bit deeper into cool features within FreeSWITCH. Since we're designing the course from scratch we can also customize it so you learn what you want to learn. So far, topics requested have focused more along the lines of specific modules and carrier-style switching services, so we're going to try and focus on those topics.

    The training is also hands-on. You'll be working with FreeSWITCH on individual workstations during the course - doing labs, etc. We'll try and keep it as interactive as we can.

And of course, if you like saving money... We have two deals going right now:
  1. If you send two or more people we are now offering a good discount. It will vary based on the number of people you send. Please contact me directly if you are interested in this option.
  2. If you sign up for the Training + ClueCon option you save ~$300! Flood yourself with FreeSWITCH knowledge and contacts by attending both events. This is a really good deal!

    For more information, please visit the website at http://www.voipkb.com/ or drop me a note.

    Thanks much and I look forward to seeing you at the training!

Sincerely,
    Darren Schreiber


From frank at impactfax.com  Sat May 29 13:04:41 2010
From: frank at impactfax.com (Frank @ Impact)
Date: Sat, 29 May 2010 16:04:41 -0400
Subject: [Freeswitch-users] how to get session of bridged Bleg
In-Reply-To: <38878699-4260-458F-839C-20084B5BEE6C@freeswitch.org>
Message-ID: <D90768A8C2D1414CA74DC9878AEED3C5@ws4>

Ok.  so after the bridge is done and the channels are hung up, how do I
know which leg disconnected and why? Who hung up first and send the bye?
 
I am initiating the bridge with
session:execute("bridge","{route=sofia/gateway/US/15555551212}
sofia/gateway/US/15555551212")
 
 
-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Brian West
Sent: Saturday, May 29, 2010 1:44 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] how to get session of bridged Bleg
 
you can't till the channel is hung up.
 
/b

 
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From saeedahmad1981 at gmail.com  Sat May 29 13:44:25 2010
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Sat, 29 May 2010 22:44:25 +0200
Subject: [Freeswitch-users] FS as Media Gateway Only
In-Reply-To: <BE75B7EE-B77D-4710-899E-AE9E771A28F7@gmail.com>
References: <AANLkTimgKRtL06xmyjXUPnpNIjlYkHC6BytzmTqKmVm7@mail.gmail.com> 
	<AANLkTikoxsXFuYBxsXXtB26Y9_8f1I1wGmE-u-I_l9PP@mail.gmail.com> 
	<AANLkTikgol1TigK9G9vvk90ogtjqeJ3nwPXqdFUXcV5W@mail.gmail.com> 
	<BE75B7EE-B77D-4710-899E-AE9E771A28F7@gmail.com>
Message-ID: <AANLkTimQUri22GlLok8z8LGSFRmufCIXdWGdA4NStbfI@mail.gmail.com>

If i understood correctly, Vitalie solutions is still workable, (although
what Code mentioned, would be ideal), because from customer side its normal
to provide multiple IPs or in most cases a whole subnet range, and call can
come any IP from theatrange, a good example is Arbinet.

Commercial SBCs like nextone support it.

Vitalie, i've a concern that in your solution how would we deal with cdrs?

Thanks

On Fri, May 28, 2010 at 1:31 PM, David Ponzone <david.ponzone at gmail.com>wrote:

> Code,
>
> you're totally right.
> In this model (FS), the media server will also be in the SIP Path.
> That's why I answered in the first place that this was not achievable with
> FS, because your idea was more a Kamaillo/RTPProxy setup, where the
> mediaserver only does RTP with the endpoints, and is not in the SIP path at
> all:
>
> inbound <--------SIP------ SIP Server/Proxy ------------SIP-------> Carrier
>                                                   |
>              <---------RTP------ MediaServer--------RTP--------------->
>
>
> Verizon Business (in Europe at least) has a such infrastrucutre: OpenSER
> for the SIP part, and Nortel GWs for the RTP.
> This way, they just give me the IPs of their OpenSER servers, and they can
> deploy as many media servers as they need without telling us (of course, we
> dont filter that).
>
> I don't know how this is implemented in Kamaillo/OpenSER but perhaps, for a
> nice bounty, that would be something possible in FS.
>
> David Ponzone  Direction Technique
>  email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 28/05/2010 ? 05:34, Code Ghar a ?crit :
>
> Hi Vitalie
>
> Thanks for providing the link and details. If I understood correctly, the
> chain of signaling would be Inbound -> FSSIP -> FSRTP -> Outbound (using
> names and terms in my original question), while the chain of media would be
> Inbound -> FSRTP -> Outbound. This way we can have multiple servers handling
> media and minimal servers handling signaling.
>
> Let me clarify a little more my motivation for asking this question in the
> first place. I work with telecom carriers on a daily basis and have seen
> many different architectures. The first biggest problem is how to load
> balance SIP traffic when you are receiving calls, if one server is not
> enough. The second biggest problem is handling all RTP, including
> transcoding. With this architecture, one or two IPs for signaling can be
> handled by most carriers. So you can beef up your hardware for signaling and
> depend less on your carrier's ability to load balance traffic for you. If
> they can do round-robin or failover for two IPs, you are golden. And then
> you can deploy multiple nodes to handle all RTP duties, without having to
> concern your carrier about multiple servers and IPs. But there's one thing
> still missing. Your outbound carrier still needs to allow traffic from
> multiple IPs because now they are dealing with FSRTP instead of FSSIP.
>
> Is there such a solution possible using FS that all signaling, for both
> inbound and outbound carriers, can be handled by a couple of FSSIP nodes
> (depending on the amount of traffic, maybe a few more) while delegating
> media responsibilities to multiple FSRTP nodes? In this situation, SIP IP is
> always, say 10.10.10.1 or 10.10.10.2, but the SDP may use media IPs
> 10.10.10.3, 10.10.10.4, 10.10.10.5, and so on. Almost all carriers I have
> seen concern themselves only with which SIP IPs they should allow and don't
> care how many or which media IPs they have to deal with. Any ideas on
> minimizing signaling IPs while adding more media IPs as required?
>
> I have seen re-invite being used in production where you can just let your
> inbound and outbound handle media between them on their own without it going
> through your network. But there are circumstances where people might need to
> pass media through their own network, chiefly to perform transcoding and
> also to handle other interop issues. It is because of this use case,
> combined with the need for minimizing signaling IPs, that the original
> question was asked.
>
>
>
>
> On Thu, May 27, 2010 at 1:15 AM, Vitalii Colosov <vetali100 at gmail.com>wrote:
>
>> Hi Code,
>>
>> I have working example doing exactly what you've described.
>> One signalling FS bridges incoming call to a set of media servers
>> (depending on ip, but you can implement any routing logic including round
>> robin) and then transfers media stream after bridging to that media server.
>>
>> You can achieve this on signalling FS by creating a Lua script that
>> contains the following lines:
>>
>> media_server="my_media_X.mydomain.com"; --to be determined by routing
>> logic
>> forwarding_session = "sofia/external/"..called_number.."@"..media_server;
>> session:setVariable("bypass_media_after_bridge", "true");
>> session:setVariable("hangup_after_bridge", "true");
>> session:execute("bridge",forwarding_session);
>>
>> The call will arrive to the selected media server successfully and media
>> stream will start bypassing signalling FS after bridge.
>>
>> You can read the following thread, it describes how you can setup such
>> configuration.
>>
>> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-March/055231.html
>>
>> I think it will fit your needs.
>>
>> Regards,
>> Vitalie
>>
>>
>> 2010/5/27 Code Ghar <codeghar at gmail.com>
>>
>>> Is it possible -- and are there any case studies, practical experience,
>>> etc -- on deploying FreeSWITCH (FS) in this architecture: one server (FSSIP)
>>> handles SIP signaling only, and multiple servers (FSRTP1, FSRTP2, ...,
>>> FSRTPn) handle all media responsibilities? So when a call comes in, the SDP
>>> contains IP of, say FSRTP1, as media handler. For this to work, FSSIP would
>>> request FSRTPx for media resources for each new call and add its IP and port
>>> in SDP. The media servers/gateways would play IVR, etc.; collect DTMF and
>>> forward as appropriate to FSSIP; perform transcoding; etc.; all while FSSIP
>>> only deals with signaling. This way multiple servers could be deployed to
>>> handle media responsibilities and only a handful would be required for
>>> signaling. In future if there's a greater need for transcoding, etc. all you
>>> need to do is deploy a media server and not have to add servers for
>>> signaling.
>>>
>>> This idea came to me because I have come across two proprietary
>>> applications that do it this way. They have a SIP component and a media
>>> component. You can run both on the same physical machine or you can separate
>>> them out into multiple machines.
>>>
>>> Another way for this could be to integrate FS as a media component to
>>> another application's SIP component. A mix-and-match, so to speak.
>>>
>>> On the flip side, deploy FS as a SIP server and use media capabilities of
>>> some other hardware or software application. For example, FS handles
>>> signaling and use dedicated hardware for media. A good example of this is
>>> illustrated (somewhat) by an image on Sangoma's website:
>>> http://www.sangoma.com/assets/images/content/transcoding_diagram.jpg.
>>> Look at the "pooled transcoding".
>>>
>>> Is FS even designed to be this modular? If so, how can the aforementioned
>>> scenario(s) be achieved?
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From vetali100 at gmail.com  Sat May 29 13:55:12 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Sat, 29 May 2010 23:55:12 +0300
Subject: [Freeswitch-users] FS as Media Gateway Only
In-Reply-To: <AANLkTimQUri22GlLok8z8LGSFRmufCIXdWGdA4NStbfI@mail.gmail.com>
References: <AANLkTimgKRtL06xmyjXUPnpNIjlYkHC6BytzmTqKmVm7@mail.gmail.com>
	<AANLkTikoxsXFuYBxsXXtB26Y9_8f1I1wGmE-u-I_l9PP@mail.gmail.com>
	<AANLkTikgol1TigK9G9vvk90ogtjqeJ3nwPXqdFUXcV5W@mail.gmail.com>
	<BE75B7EE-B77D-4710-899E-AE9E771A28F7@gmail.com>
	<AANLkTimQUri22GlLok8z8LGSFRmufCIXdWGdA4NStbfI@mail.gmail.com>
Message-ID: <AANLkTimI28Pi_oRNRkgNtT4dT9_cLTPJ9a2XVkgGFzAW@mail.gmail.com>

In our implementation only Media servers generate CDRs. No Media session -
no CDR (at least we need exactly this behavior).

However if you need, you can generate CDRs only on Signalling servers I
beleive.

Regards,
Vitalie

2010/5/29 Saeed Ahmed <saeedahmad1981 at gmail.com>

> If i understood correctly, Vitalie solutions is still workable, (although
> what Code mentioned, would be ideal), because from customer side its normal
> to provide multiple IPs or in most cases a whole subnet range, and call can
> come any IP from theatrange, a good example is Arbinet.
>
> Commercial SBCs like nextone support it.
>
> Vitalie, i've a concern that in your solution how would we deal with cdrs?
>
> Thanks
>
> On Fri, May 28, 2010 at 1:31 PM, David Ponzone <david.ponzone at gmail.com>wrote:
>
>> Code,
>>
>> you're totally right.
>> In this model (FS), the media server will also be in the SIP Path.
>> That's why I answered in the first place that this was not achievable with
>> FS, because your idea was more a Kamaillo/RTPProxy setup, where the
>> mediaserver only does RTP with the endpoints, and is not in the SIP path at
>> all:
>>
>> inbound <--------SIP------ SIP Server/Proxy ------------SIP------->
>> Carrier
>>                                                   |
>>              <---------RTP------ MediaServer--------RTP--------------->
>>
>>
>> Verizon Business (in Europe at least) has a such infrastrucutre: OpenSER
>> for the SIP part, and Nortel GWs for the RTP.
>> This way, they just give me the IPs of their OpenSER servers, and they can
>> deploy as many media servers as they need without telling us (of course, we
>> dont filter that).
>>
>> I don't know how this is implemented in Kamaillo/OpenSER but perhaps, for
>> a nice bounty, that would be something possible in FS.
>>
>> David Ponzone  Direction Technique
>>  email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 28/05/2010 ? 05:34, Code Ghar a ?crit :
>>
>> Hi Vitalie
>>
>> Thanks for providing the link and details. If I understood correctly, the
>> chain of signaling would be Inbound -> FSSIP -> FSRTP -> Outbound (using
>> names and terms in my original question), while the chain of media would be
>> Inbound -> FSRTP -> Outbound. This way we can have multiple servers handling
>> media and minimal servers handling signaling.
>>
>> Let me clarify a little more my motivation for asking this question in the
>> first place. I work with telecom carriers on a daily basis and have seen
>> many different architectures. The first biggest problem is how to load
>> balance SIP traffic when you are receiving calls, if one server is not
>> enough. The second biggest problem is handling all RTP, including
>> transcoding. With this architecture, one or two IPs for signaling can be
>> handled by most carriers. So you can beef up your hardware for signaling and
>> depend less on your carrier's ability to load balance traffic for you. If
>> they can do round-robin or failover for two IPs, you are golden. And then
>> you can deploy multiple nodes to handle all RTP duties, without having to
>> concern your carrier about multiple servers and IPs. But there's one thing
>> still missing. Your outbound carrier still needs to allow traffic from
>> multiple IPs because now they are dealing with FSRTP instead of FSSIP.
>>
>> Is there such a solution possible using FS that all signaling, for both
>> inbound and outbound carriers, can be handled by a couple of FSSIP nodes
>> (depending on the amount of traffic, maybe a few more) while delegating
>> media responsibilities to multiple FSRTP nodes? In this situation, SIP IP is
>> always, say 10.10.10.1 or 10.10.10.2, but the SDP may use media IPs
>> 10.10.10.3, 10.10.10.4, 10.10.10.5, and so on. Almost all carriers I have
>> seen concern themselves only with which SIP IPs they should allow and don't
>> care how many or which media IPs they have to deal with. Any ideas on
>> minimizing signaling IPs while adding more media IPs as required?
>>
>> I have seen re-invite being used in production where you can just let your
>> inbound and outbound handle media between them on their own without it going
>> through your network. But there are circumstances where people might need to
>> pass media through their own network, chiefly to perform transcoding and
>> also to handle other interop issues. It is because of this use case,
>> combined with the need for minimizing signaling IPs, that the original
>> question was asked.
>>
>>
>>
>>
>> On Thu, May 27, 2010 at 1:15 AM, Vitalii Colosov <vetali100 at gmail.com>wrote:
>>
>>> Hi Code,
>>>
>>> I have working example doing exactly what you've described.
>>> One signalling FS bridges incoming call to a set of media servers
>>> (depending on ip, but you can implement any routing logic including round
>>> robin) and then transfers media stream after bridging to that media server.
>>>
>>> You can achieve this on signalling FS by creating a Lua script that
>>> contains the following lines:
>>>
>>> media_server="my_media_X.mydomain.com"; --to be determined by routing
>>> logic
>>> forwarding_session = "sofia/external/"..called_number.."@"..media_server;
>>> session:setVariable("bypass_media_after_bridge", "true");
>>> session:setVariable("hangup_after_bridge", "true");
>>> session:execute("bridge",forwarding_session);
>>>
>>> The call will arrive to the selected media server successfully and media
>>> stream will start bypassing signalling FS after bridge.
>>>
>>> You can read the following thread, it describes how you can setup such
>>> configuration.
>>>
>>> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-March/055231.html
>>>
>>> I think it will fit your needs.
>>>
>>> Regards,
>>> Vitalie
>>>
>>>
>>> 2010/5/27 Code Ghar <codeghar at gmail.com>
>>>
>>>> Is it possible -- and are there any case studies, practical experience,
>>>> etc -- on deploying FreeSWITCH (FS) in this architecture: one server (FSSIP)
>>>> handles SIP signaling only, and multiple servers (FSRTP1, FSRTP2, ...,
>>>> FSRTPn) handle all media responsibilities? So when a call comes in, the SDP
>>>> contains IP of, say FSRTP1, as media handler. For this to work, FSSIP would
>>>> request FSRTPx for media resources for each new call and add its IP and port
>>>> in SDP. The media servers/gateways would play IVR, etc.; collect DTMF and
>>>> forward as appropriate to FSSIP; perform transcoding; etc.; all while FSSIP
>>>> only deals with signaling. This way multiple servers could be deployed to
>>>> handle media responsibilities and only a handful would be required for
>>>> signaling. In future if there's a greater need for transcoding, etc. all you
>>>> need to do is deploy a media server and not have to add servers for
>>>> signaling.
>>>>
>>>> This idea came to me because I have come across two proprietary
>>>> applications that do it this way. They have a SIP component and a media
>>>> component. You can run both on the same physical machine or you can separate
>>>> them out into multiple machines.
>>>>
>>>> Another way for this could be to integrate FS as a media component to
>>>> another application's SIP component. A mix-and-match, so to speak.
>>>>
>>>> On the flip side, deploy FS as a SIP server and use media capabilities
>>>> of some other hardware or software application. For example, FS handles
>>>> signaling and use dedicated hardware for media. A good example of this is
>>>> illustrated (somewhat) by an image on Sangoma's website:
>>>> http://www.sangoma.com/assets/images/content/transcoding_diagram.jpg.
>>>> Look at the "pooled transcoding".
>>>>
>>>> Is FS even designed to be this modular? If so, how can the
>>>> aforementioned scenario(s) be achieved?
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From babak.freeswitch at gmail.com  Sat May 29 22:09:45 2010
From: babak.freeswitch at gmail.com (babak yakhchali)
Date: Sun, 30 May 2010 09:39:45 +0430
Subject: [Freeswitch-users] nat problem!
In-Reply-To: <AANLkTil6xFDgtEH10rr353o2aJUiFmM311mMgDD6JhIu@mail.gmail.com>
References: <AANLkTin2PsoyY9jqaq2CW6EY-IfyyUxSUQFPnHBGaQuA@mail.gmail.com>
	<880247BE-772D-4595-B591-D0577D9F9756@freeswitch.org>
	<AANLkTinthmE7uLkV4g_zcuWYiyBQ4wiDf8zdXaUnpW5M@mail.gmail.com>
	<AANLkTimG25bRUAlc8KwB8v9XjQM7xqxbnnT5OnE7mC34@mail.gmail.com>
	<F1C6AB5C-545A-487F-B660-4168A1E16219@gmail.com>
	<AANLkTikBTfJbSaWLA_7EUMNPBYlNZbkPEKLbJF8y64Pm@mail.gmail.com>
	<AANLkTilNIDxTurbL7CGnUvMjkbdDi2kSZD2EhDaodP-q@mail.gmail.com>
	<BEF3111D-C78F-4BF3-9C26-D066EC128A66@gmail.com>
	<AANLkTikhUnNW9RfnZxKnLzV_hUCJuNhnFnYLaF1inw0L@mail.gmail.com>
	<D2590FAD-3115-40D3-BD0F-268F1E5A4846@freeswitch.org>
	<AANLkTimnubtYMhVvSx3seX5tipF-Vcy_2nbuFay06r_3@mail.gmail.com>
	<AANLkTil6xFDgtEH10rr353o2aJUiFmM311mMgDD6JhIu@mail.gmail.com>
Message-ID: <AANLkTin2p_PR0DBw6dnx59-DGMlsUDj7BEAuAsPY9N6_@mail.gmail.com>

Hi all
thanx for ur answers. problem solved (thanx to stkn on IRC) by disabling
apply-nat-acl and setting external ip for sip and rtp to local-ipv4 (I think
the acl was the problem).
thank u all
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From kees at mroffice.org  Sat May 29 23:04:23 2010
From: kees at mroffice.org (Kees Varekamp)
Date: Sun, 30 May 2010 18:04:23 +1200
Subject: [Freeswitch-users] recording to ogg vorbis
Message-ID: <AANLkTine9yRZCLFMepJqQozZhdXJ9NXI6QYmaLsHWRId@mail.gmail.com>

Hello,

Can I record to ogg vorbis?

I've read that anything other than wav is not recommended, but the good
thing about ogg is that it is the only format that can be played in all the
html5 compliant browsers.

If it can't be done, is there any way to build a conversion hook into the
record command? (convert after finished recording)

Thanks,

Kees
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From brian at freeswitch.org  Sat May 29 23:26:24 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 30 May 2010 01:26:24 -0500
Subject: [Freeswitch-users] recording to ogg vorbis
In-Reply-To: <AANLkTine9yRZCLFMepJqQozZhdXJ9NXI6QYmaLsHWRId@mail.gmail.com>
References: <AANLkTine9yRZCLFMepJqQozZhdXJ9NXI6QYmaLsHWRId@mail.gmail.com>
Message-ID: <D13190E0-926B-4989-9FC5-CEABD4124077@freeswitch.org>

no

Sent from my iPad

On May 30, 2010, at 1:04 AM, Kees Varekamp <kees at mroffice.org> wrote:

> Hello,
> 
> Can I record to ogg vorbis?
> 
> I've read that anything other than wav is not recommended, but the good thing about ogg is that it is the only format that can be played in all the html5 compliant browsers.
> 
> If it can't be done, is there any way to build a conversion hook into the record command? (convert after finished recording)
> 
> Thanks,
> 
> Kees
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From kees at mroffice.org  Sat May 29 23:34:31 2010
From: kees at mroffice.org (Kees Varekamp)
Date: Sun, 30 May 2010 18:34:31 +1200
Subject: [Freeswitch-users] recording to ogg vorbis
In-Reply-To: <D13190E0-926B-4989-9FC5-CEABD4124077@freeswitch.org>
References: <AANLkTine9yRZCLFMepJqQozZhdXJ9NXI6QYmaLsHWRId@mail.gmail.com>
	<D13190E0-926B-4989-9FC5-CEABD4124077@freeswitch.org>
Message-ID: <AANLkTim4OGNAtPR0v23CSZaHIdklgDBDM3HCDopPQrSf@mail.gmail.com>

OK tnx :-)

On Sun, May 30, 2010 at 18:26, Brian West <brian at freeswitch.org> wrote:

> no
>
> Sent from my iPad
>
> On May 30, 2010, at 1:04 AM, Kees Varekamp <kees at mroffice.org> wrote:
>
> > Hello,
> >
> > Can I record to ogg vorbis?
> >
> > I've read that anything other than wav is not recommended, but the good
> thing about ogg is that it is the only format that can be played in all the
> html5 compliant browsers.
> >
> > If it can't be done, is there any way to build a conversion hook into the
> record command? (convert after finished recording)
> >
> > Thanks,
> >
> > Kees
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Sat May 29 23:52:07 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 30 May 2010 01:52:07 -0500
Subject: [Freeswitch-users] recording to ogg vorbis
In-Reply-To: <AANLkTim4OGNAtPR0v23CSZaHIdklgDBDM3HCDopPQrSf@mail.gmail.com>
References: <AANLkTine9yRZCLFMepJqQozZhdXJ9NXI6QYmaLsHWRId@mail.gmail.com>
	<D13190E0-926B-4989-9FC5-CEABD4124077@freeswitch.org>
	<AANLkTim4OGNAtPR0v23CSZaHIdklgDBDM3HCDopPQrSf@mail.gmail.com>
Message-ID: <AE9F2D1E-00B0-4066-8FFC-78BBDF5ABCC4@freeswitch.org>

you could write a format module to allow that.

Sent from my iPad

On May 30, 2010, at 1:34 AM, Kees Varekamp <kees at mroffice.org> wrote:

> OK tnx :-)


From kees at mroffice.org  Sun May 30 00:24:16 2010
From: kees at mroffice.org (Kees Varekamp)
Date: Sun, 30 May 2010 19:24:16 +1200
Subject: [Freeswitch-users] recording to ogg vorbis
In-Reply-To: <AE9F2D1E-00B0-4066-8FFC-78BBDF5ABCC4@freeswitch.org>
References: <AANLkTine9yRZCLFMepJqQozZhdXJ9NXI6QYmaLsHWRId@mail.gmail.com>
	<D13190E0-926B-4989-9FC5-CEABD4124077@freeswitch.org>
	<AANLkTim4OGNAtPR0v23CSZaHIdklgDBDM3HCDopPQrSf@mail.gmail.com>
	<AE9F2D1E-00B0-4066-8FFC-78BBDF5ABCC4@freeswitch.org>
Message-ID: <AANLkTinIifgtI88Y7VdL_orE9d1xvKBFyBSWUGZ17O0C@mail.gmail.com>

I am keen but my c skills are not top notch. But I'll give it a go.

tnx,
Kees

On Sun, May 30, 2010 at 18:52, Brian West <brian at freeswitch.org> wrote:

> you could write a format module to allow that.
>
> Sent from my iPad
>
> On May 30, 2010, at 1:34 AM, Kees Varekamp <kees at mroffice.org> wrote:
>
> > OK tnx :-)
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jalsot at gmail.com  Sun May 30 02:01:35 2010
From: jalsot at gmail.com (Tamas)
Date: Sun, 30 May 2010 11:01:35 +0200
Subject: [Freeswitch-users] recording to ogg vorbis
In-Reply-To: <AANLkTinIifgtI88Y7VdL_orE9d1xvKBFyBSWUGZ17O0C@mail.gmail.com>
References: <AANLkTine9yRZCLFMepJqQozZhdXJ9NXI6QYmaLsHWRId@mail.gmail.com>	<D13190E0-926B-4989-9FC5-CEABD4124077@freeswitch.org>	<AANLkTim4OGNAtPR0v23CSZaHIdklgDBDM3HCDopPQrSf@mail.gmail.com>	<AE9F2D1E-00B0-4066-8FFC-78BBDF5ABCC4@freeswitch.org>
	<AANLkTinIifgtI88Y7VdL_orE9d1xvKBFyBSWUGZ17O0C@mail.gmail.com>
Message-ID: <4C02296F.1080407@gmail.com>

Hello,

maybe upgrading libsndfile could do the trick as new version supports
OGG/Vorbis.
(http://www.mega-nerd.com/libsndfile/).

Regards,
    Tamas

2010-05-30 09:24 keltez?ssel, Kees Varekamp ?rta:
> I am keen but my c skills are not top notch. But I'll give it a go.
>
> tnx,
> Kees
>
> On Sun, May 30, 2010 at 18:52, Brian West <brian at freeswitch.org
> <mailto:brian at freeswitch.org>> wrote:
>
>     you could write a format module to allow that.
>
>     Sent from my iPad
>
>     On May 30, 2010, at 1:34 AM, Kees Varekamp <kees at mroffice.org
>     <mailto:kees at mroffice.org>> wrote:
>
>     > OK tnx :-)
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   
   
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From saeedahmad1981 at gmail.com  Sun May 30 02:36:37 2010
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Sun, 30 May 2010 11:36:37 +0200
Subject: [Freeswitch-users] FS as Media Gateway Only
In-Reply-To: <AANLkTimI28Pi_oRNRkgNtT4dT9_cLTPJ9a2XVkgGFzAW@mail.gmail.com>
References: <AANLkTimgKRtL06xmyjXUPnpNIjlYkHC6BytzmTqKmVm7@mail.gmail.com> 
	<AANLkTikoxsXFuYBxsXXtB26Y9_8f1I1wGmE-u-I_l9PP@mail.gmail.com> 
	<AANLkTikgol1TigK9G9vvk90ogtjqeJ3nwPXqdFUXcV5W@mail.gmail.com> 
	<BE75B7EE-B77D-4710-899E-AE9E771A28F7@gmail.com>
	<AANLkTimQUri22GlLok8z8LGSFRmufCIXdWGdA4NStbfI@mail.gmail.com> 
	<AANLkTimI28Pi_oRNRkgNtT4dT9_cLTPJ9a2XVkgGFzAW@mail.gmail.com>
Message-ID: <AANLkTimBbGprWu53lQqWhaTmLRdh25l-yCmYbbT1MXBq@mail.gmail.com>

Another point, with following script:

media_server="my_media_X.mydomain.com <http://my_media_x.mydomain.com/>";
--to be determined by routing logic
forwarding_session = "sofia/external/"..called_number.."@"..media_server;
session:setVariable("bypass_media_after_bridge", "true");
session:setVariable("hangup_after_bridge", "true");
session:execute("bridge",forwarding_session);

we can bill the customer (inbound) but supplier (outbound) is behind
media_server, so i think it would be a problem to manage ,supplier side for
billing and statistics, on signalling server.

On Sat, May 29, 2010 at 10:55 PM, Vitalii Colosov <vetali100 at gmail.com>wrote:

> In our implementation only Media servers generate CDRs. No Media session -
> no CDR (at least we need exactly this behavior).
>
> However if you need, you can generate CDRs only on Signalling servers I
> beleive.
>
> Regards,
> Vitalie
>
> 2010/5/29 Saeed Ahmed <saeedahmad1981 at gmail.com>
>
> If i understood correctly, Vitalie solutions is still workable, (although
>> what Code mentioned, would be ideal), because from customer side its normal
>> to provide multiple IPs or in most cases a whole subnet range, and call can
>> come any IP from theatrange, a good example is Arbinet.
>>
>> Commercial SBCs like nextone support it.
>>
>> Vitalie, i've a concern that in your solution how would we deal with
>> cdrs?
>>
>> Thanks
>>
>> On Fri, May 28, 2010 at 1:31 PM, David Ponzone <david.ponzone at gmail.com>wrote:
>>
>>> Code,
>>>
>>> you're totally right.
>>> In this model (FS), the media server will also be in the SIP Path.
>>> That's why I answered in the first place that this was not achievable
>>> with FS, because your idea was more a Kamaillo/RTPProxy setup, where the
>>> mediaserver only does RTP with the endpoints, and is not in the SIP path at
>>> all:
>>>
>>> inbound <--------SIP------ SIP Server/Proxy ------------SIP------->
>>> Carrier
>>>                                                   |
>>>              <---------RTP------ MediaServer--------RTP--------------->
>>>
>>>
>>> Verizon Business (in Europe at least) has a such infrastrucutre: OpenSER
>>> for the SIP part, and Nortel GWs for the RTP.
>>> This way, they just give me the IPs of their OpenSER servers, and they
>>> can deploy as many media servers as they need without telling us (of course,
>>> we dont filter that).
>>>
>>> I don't know how this is implemented in Kamaillo/OpenSER but perhaps, for
>>> a nice bounty, that would be something possible in FS.
>>>
>>> David Ponzone  Direction Technique
>>>  email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>
>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>> l'exp?diteur.*
>>> *
>>> *
>>>
>>>
>>>
>>> Le 28/05/2010 ? 05:34, Code Ghar a ?crit :
>>>
>>> Hi Vitalie
>>>
>>> Thanks for providing the link and details. If I understood correctly, the
>>> chain of signaling would be Inbound -> FSSIP -> FSRTP -> Outbound (using
>>> names and terms in my original question), while the chain of media would be
>>> Inbound -> FSRTP -> Outbound. This way we can have multiple servers handling
>>> media and minimal servers handling signaling.
>>>
>>> Let me clarify a little more my motivation for asking this question in
>>> the first place. I work with telecom carriers on a daily basis and have seen
>>> many different architectures. The first biggest problem is how to load
>>> balance SIP traffic when you are receiving calls, if one server is not
>>> enough. The second biggest problem is handling all RTP, including
>>> transcoding. With this architecture, one or two IPs for signaling can be
>>> handled by most carriers. So you can beef up your hardware for signaling and
>>> depend less on your carrier's ability to load balance traffic for you. If
>>> they can do round-robin or failover for two IPs, you are golden. And then
>>> you can deploy multiple nodes to handle all RTP duties, without having to
>>> concern your carrier about multiple servers and IPs. But there's one thing
>>> still missing. Your outbound carrier still needs to allow traffic from
>>> multiple IPs because now they are dealing with FSRTP instead of FSSIP.
>>>
>>> Is there such a solution possible using FS that all signaling, for both
>>> inbound and outbound carriers, can be handled by a couple of FSSIP nodes
>>> (depending on the amount of traffic, maybe a few more) while delegating
>>> media responsibilities to multiple FSRTP nodes? In this situation, SIP IP is
>>> always, say 10.10.10.1 or 10.10.10.2, but the SDP may use media IPs
>>> 10.10.10.3, 10.10.10.4, 10.10.10.5, and so on. Almost all carriers I have
>>> seen concern themselves only with which SIP IPs they should allow and don't
>>> care how many or which media IPs they have to deal with. Any ideas on
>>> minimizing signaling IPs while adding more media IPs as required?
>>>
>>> I have seen re-invite being used in production where you can just let
>>> your inbound and outbound handle media between them on their own without it
>>> going through your network. But there are circumstances where people might
>>> need to pass media through their own network, chiefly to perform transcoding
>>> and also to handle other interop issues. It is because of this use case,
>>> combined with the need for minimizing signaling IPs, that the original
>>> question was asked.
>>>
>>>
>>>
>>>
>>> On Thu, May 27, 2010 at 1:15 AM, Vitalii Colosov <vetali100 at gmail.com>wrote:
>>>
>>>> Hi Code,
>>>>
>>>> I have working example doing exactly what you've described.
>>>> One signalling FS bridges incoming call to a set of media servers
>>>> (depending on ip, but you can implement any routing logic including round
>>>> robin) and then transfers media stream after bridging to that media server.
>>>>
>>>> You can achieve this on signalling FS by creating a Lua script that
>>>> contains the following lines:
>>>>
>>>> media_server="my_media_X.mydomain.com"; --to be determined by routing
>>>> logic
>>>> forwarding_session =
>>>> "sofia/external/"..called_number.."@"..media_server;
>>>> session:setVariable("bypass_media_after_bridge", "true");
>>>> session:setVariable("hangup_after_bridge", "true");
>>>> session:execute("bridge",forwarding_session);
>>>>
>>>> The call will arrive to the selected media server successfully and media
>>>> stream will start bypassing signalling FS after bridge.
>>>>
>>>> You can read the following thread, it describes how you can setup such
>>>> configuration.
>>>>
>>>> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-March/055231.html
>>>>
>>>> I think it will fit your needs.
>>>>
>>>> Regards,
>>>> Vitalie
>>>>
>>>>
>>>> 2010/5/27 Code Ghar <codeghar at gmail.com>
>>>>
>>>>> Is it possible -- and are there any case studies, practical experience,
>>>>> etc -- on deploying FreeSWITCH (FS) in this architecture: one server (FSSIP)
>>>>> handles SIP signaling only, and multiple servers (FSRTP1, FSRTP2, ...,
>>>>> FSRTPn) handle all media responsibilities? So when a call comes in, the SDP
>>>>> contains IP of, say FSRTP1, as media handler. For this to work, FSSIP would
>>>>> request FSRTPx for media resources for each new call and add its IP and port
>>>>> in SDP. The media servers/gateways would play IVR, etc.; collect DTMF and
>>>>> forward as appropriate to FSSIP; perform transcoding; etc.; all while FSSIP
>>>>> only deals with signaling. This way multiple servers could be deployed to
>>>>> handle media responsibilities and only a handful would be required for
>>>>> signaling. In future if there's a greater need for transcoding, etc. all you
>>>>> need to do is deploy a media server and not have to add servers for
>>>>> signaling.
>>>>>
>>>>> This idea came to me because I have come across two proprietary
>>>>> applications that do it this way. They have a SIP component and a media
>>>>> component. You can run both on the same physical machine or you can separate
>>>>> them out into multiple machines.
>>>>>
>>>>> Another way for this could be to integrate FS as a media component to
>>>>> another application's SIP component. A mix-and-match, so to speak.
>>>>>
>>>>> On the flip side, deploy FS as a SIP server and use media capabilities
>>>>> of some other hardware or software application. For example, FS handles
>>>>> signaling and use dedicated hardware for media. A good example of this is
>>>>> illustrated (somewhat) by an image on Sangoma's website:
>>>>> http://www.sangoma.com/assets/images/content/transcoding_diagram.jpg.
>>>>> Look at the "pooled transcoding".
>>>>>
>>>>> Is FS even designed to be this modular? If so, how can the
>>>>> aforementioned scenario(s) be achieved?
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From vetali100 at gmail.com  Sun May 30 02:45:37 2010
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Sun, 30 May 2010 12:45:37 +0300
Subject: [Freeswitch-users] FS as Media Gateway Only
In-Reply-To: <AANLkTimBbGprWu53lQqWhaTmLRdh25l-yCmYbbT1MXBq@mail.gmail.com>
References: <AANLkTimgKRtL06xmyjXUPnpNIjlYkHC6BytzmTqKmVm7@mail.gmail.com>
	<AANLkTikoxsXFuYBxsXXtB26Y9_8f1I1wGmE-u-I_l9PP@mail.gmail.com>
	<AANLkTikgol1TigK9G9vvk90ogtjqeJ3nwPXqdFUXcV5W@mail.gmail.com>
	<BE75B7EE-B77D-4710-899E-AE9E771A28F7@gmail.com>
	<AANLkTimQUri22GlLok8z8LGSFRmufCIXdWGdA4NStbfI@mail.gmail.com>
	<AANLkTimI28Pi_oRNRkgNtT4dT9_cLTPJ9a2XVkgGFzAW@mail.gmail.com>
	<AANLkTimBbGprWu53lQqWhaTmLRdh25l-yCmYbbT1MXBq@mail.gmail.com>
Message-ID: <AANLkTimSpVOYVB2BiKZWVr3dNrjy0ufQv7YsqAAbdRxS@mail.gmail.com>

Could you please provide an example of what exactly do you think is a
problem?
Because now I don't have any problem with this configuration and I am
wondering what is the exact scenario where we cannot use such configuration.

Thank you,
Vitalie

2010/5/30 Saeed Ahmed <saeedahmad1981 at gmail.com>

> Another point, with following script:
>
> media_server="my_media_X.mydomain.com <http://my_media_x.mydomain.com/>";
> --to be determined by routing logic
> forwarding_session = "sofia/external/"..called_number.."@"..media_server;
> session:setVariable("bypass_media_after_bridge", "true");
> session:setVariable("hangup_after_bridge", "true");
> session:execute("bridge",forwarding_session);
>
> we can bill the customer (inbound) but supplier (outbound) is behind
> media_server, so i think it would be a problem to manage ,supplier side for
> billing and statistics, on signalling server.
>
> On Sat, May 29, 2010 at 10:55 PM, Vitalii Colosov <vetali100 at gmail.com>wrote:
>
>> In our implementation only Media servers generate CDRs. No Media session -
>> no CDR (at least we need exactly this behavior).
>>
>> However if you need, you can generate CDRs only on Signalling servers I
>> beleive.
>>
>> Regards,
>> Vitalie
>>
>> 2010/5/29 Saeed Ahmed <saeedahmad1981 at gmail.com>
>>
>> If i understood correctly, Vitalie solutions is still workable, (although
>>> what Code mentioned, would be ideal), because from customer side its normal
>>> to provide multiple IPs or in most cases a whole subnet range, and call can
>>> come any IP from theatrange, a good example is Arbinet.
>>>
>>> Commercial SBCs like nextone support it.
>>>
>>> Vitalie, i've a concern that in your solution how would we deal with
>>> cdrs?
>>>
>>> Thanks
>>>
>>> On Fri, May 28, 2010 at 1:31 PM, David Ponzone <david.ponzone at gmail.com>wrote:
>>>
>>>> Code,
>>>>
>>>> you're totally right.
>>>> In this model (FS), the media server will also be in the SIP Path.
>>>> That's why I answered in the first place that this was not achievable
>>>> with FS, because your idea was more a Kamaillo/RTPProxy setup, where the
>>>> mediaserver only does RTP with the endpoints, and is not in the SIP path at
>>>> all:
>>>>
>>>> inbound <--------SIP------ SIP Server/Proxy ------------SIP------->
>>>> Carrier
>>>>                                                   |
>>>>              <---------RTP------ MediaServer--------RTP--------------->
>>>>
>>>>
>>>> Verizon Business (in Europe at least) has a such infrastrucutre: OpenSER
>>>> for the SIP part, and Nortel GWs for the RTP.
>>>> This way, they just give me the IPs of their OpenSER servers, and they
>>>> can deploy as many media servers as they need without telling us (of course,
>>>> we dont filter that).
>>>>
>>>> I don't know how this is implemented in Kamaillo/OpenSER but perhaps,
>>>> for a nice bounty, that would be something possible in FS.
>>>>
>>>> David Ponzone  Direction Technique
>>>>  email: david.ponzone at ipeva.fr
>>>> tel:      01 74 03 18 97
>>>> gsm:   06 66 98 76 34
>>>>
>>>> Service Client IPeva
>>>> tel:      0811 46 26 26
>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>>
>>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
>>>> ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>>> l'exp?diteur.*
>>>> *
>>>> *
>>>>
>>>>
>>>>
>>>> Le 28/05/2010 ? 05:34, Code Ghar a ?crit :
>>>>
>>>> Hi Vitalie
>>>>
>>>> Thanks for providing the link and details. If I understood correctly,
>>>> the chain of signaling would be Inbound -> FSSIP -> FSRTP -> Outbound (using
>>>> names and terms in my original question), while the chain of media would be
>>>> Inbound -> FSRTP -> Outbound. This way we can have multiple servers handling
>>>> media and minimal servers handling signaling.
>>>>
>>>> Let me clarify a little more my motivation for asking this question in
>>>> the first place. I work with telecom carriers on a daily basis and have seen
>>>> many different architectures. The first biggest problem is how to load
>>>> balance SIP traffic when you are receiving calls, if one server is not
>>>> enough. The second biggest problem is handling all RTP, including
>>>> transcoding. With this architecture, one or two IPs for signaling can be
>>>> handled by most carriers. So you can beef up your hardware for signaling and
>>>> depend less on your carrier's ability to load balance traffic for you. If
>>>> they can do round-robin or failover for two IPs, you are golden. And then
>>>> you can deploy multiple nodes to handle all RTP duties, without having to
>>>> concern your carrier about multiple servers and IPs. But there's one thing
>>>> still missing. Your outbound carrier still needs to allow traffic from
>>>> multiple IPs because now they are dealing with FSRTP instead of FSSIP.
>>>>
>>>> Is there such a solution possible using FS that all signaling, for both
>>>> inbound and outbound carriers, can be handled by a couple of FSSIP nodes
>>>> (depending on the amount of traffic, maybe a few more) while delegating
>>>> media responsibilities to multiple FSRTP nodes? In this situation, SIP IP is
>>>> always, say 10.10.10.1 or 10.10.10.2, but the SDP may use media IPs
>>>> 10.10.10.3, 10.10.10.4, 10.10.10.5, and so on. Almost all carriers I have
>>>> seen concern themselves only with which SIP IPs they should allow and don't
>>>> care how many or which media IPs they have to deal with. Any ideas on
>>>> minimizing signaling IPs while adding more media IPs as required?
>>>>
>>>> I have seen re-invite being used in production where you can just let
>>>> your inbound and outbound handle media between them on their own without it
>>>> going through your network. But there are circumstances where people might
>>>> need to pass media through their own network, chiefly to perform transcoding
>>>> and also to handle other interop issues. It is because of this use case,
>>>> combined with the need for minimizing signaling IPs, that the original
>>>> question was asked.
>>>>
>>>>
>>>>
>>>>
>>>> On Thu, May 27, 2010 at 1:15 AM, Vitalii Colosov <vetali100 at gmail.com>wrote:
>>>>
>>>>> Hi Code,
>>>>>
>>>>> I have working example doing exactly what you've described.
>>>>> One signalling FS bridges incoming call to a set of media servers
>>>>> (depending on ip, but you can implement any routing logic including round
>>>>> robin) and then transfers media stream after bridging to that media server.
>>>>>
>>>>> You can achieve this on signalling FS by creating a Lua script that
>>>>> contains the following lines:
>>>>>
>>>>> media_server="my_media_X.mydomain.com"; --to be determined by routing
>>>>> logic
>>>>> forwarding_session =
>>>>> "sofia/external/"..called_number.."@"..media_server;
>>>>> session:setVariable("bypass_media_after_bridge", "true");
>>>>> session:setVariable("hangup_after_bridge", "true");
>>>>> session:execute("bridge",forwarding_session);
>>>>>
>>>>> The call will arrive to the selected media server successfully and
>>>>> media stream will start bypassing signalling FS after bridge.
>>>>>
>>>>> You can read the following thread, it describes how you can setup such
>>>>> configuration.
>>>>>
>>>>> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-March/055231.html
>>>>>
>>>>> I think it will fit your needs.
>>>>>
>>>>> Regards,
>>>>> Vitalie
>>>>>
>>>>>
>>>>> 2010/5/27 Code Ghar <codeghar at gmail.com>
>>>>>
>>>>>> Is it possible -- and are there any case studies, practical
>>>>>> experience, etc -- on deploying FreeSWITCH (FS) in this architecture: one
>>>>>> server (FSSIP) handles SIP signaling only, and multiple servers (FSRTP1,
>>>>>> FSRTP2, ..., FSRTPn) handle all media responsibilities? So when a call comes
>>>>>> in, the SDP contains IP of, say FSRTP1, as media handler. For this to work,
>>>>>> FSSIP would request FSRTPx for media resources for each new call and add its
>>>>>> IP and port in SDP. The media servers/gateways would play IVR, etc.; collect
>>>>>> DTMF and forward as appropriate to FSSIP; perform transcoding; etc.; all
>>>>>> while FSSIP only deals with signaling. This way multiple servers could be
>>>>>> deployed to handle media responsibilities and only a handful would be
>>>>>> required for signaling. In future if there's a greater need for transcoding,
>>>>>> etc. all you need to do is deploy a media server and not have to add servers
>>>>>> for signaling.
>>>>>>
>>>>>> This idea came to me because I have come across two proprietary
>>>>>> applications that do it this way. They have a SIP component and a media
>>>>>> component. You can run both on the same physical machine or you can separate
>>>>>> them out into multiple machines.
>>>>>>
>>>>>> Another way for this could be to integrate FS as a media component to
>>>>>> another application's SIP component. A mix-and-match, so to speak.
>>>>>>
>>>>>> On the flip side, deploy FS as a SIP server and use media capabilities
>>>>>> of some other hardware or software application. For example, FS handles
>>>>>> signaling and use dedicated hardware for media. A good example of this is
>>>>>> illustrated (somewhat) by an image on Sangoma's website:
>>>>>> http://www.sangoma.com/assets/images/content/transcoding_diagram.jpg.
>>>>>> Look at the "pooled transcoding".
>>>>>>
>>>>>> Is FS even designed to be this modular? If so, how can the
>>>>>> aforementioned scenario(s) be achieved?
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From saeedahmad1981 at gmail.com  Sun May 30 04:21:10 2010
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Sun, 30 May 2010 13:21:10 +0200
Subject: [Freeswitch-users] FS as Media Gateway Only
In-Reply-To: <AANLkTimSpVOYVB2BiKZWVr3dNrjy0ufQv7YsqAAbdRxS@mail.gmail.com>
References: <AANLkTimgKRtL06xmyjXUPnpNIjlYkHC6BytzmTqKmVm7@mail.gmail.com> 
	<AANLkTikoxsXFuYBxsXXtB26Y9_8f1I1wGmE-u-I_l9PP@mail.gmail.com> 
	<AANLkTikgol1TigK9G9vvk90ogtjqeJ3nwPXqdFUXcV5W@mail.gmail.com> 
	<BE75B7EE-B77D-4710-899E-AE9E771A28F7@gmail.com>
	<AANLkTimQUri22GlLok8z8LGSFRmufCIXdWGdA4NStbfI@mail.gmail.com> 
	<AANLkTimI28Pi_oRNRkgNtT4dT9_cLTPJ9a2XVkgGFzAW@mail.gmail.com> 
	<AANLkTimBbGprWu53lQqWhaTmLRdh25l-yCmYbbT1MXBq@mail.gmail.com> 
	<AANLkTimSpVOYVB2BiKZWVr3dNrjy0ufQv7YsqAAbdRxS@mail.gmail.com>
Message-ID: <AANLkTim5o5dc84N4Ag9PenNKvVPfrfr2cQWIQmjVhWuT@mail.gmail.com>

Hi,

Actually i am trying to setup the environment to test it real time,
but theoretically it looks like that:


               --> MS_1 ------> Supplier (outbound)
Inbound ---->  RS  --> MS_2 --> Supplier (outbound)
               --> MS_N ------> Supplier (outbound)

So what i would like to do, when call hits RS, it will look in
database (using curl or lua etc..) for end supplier (outbound) and
also selects MS_N (to deal RTP media), so that on a central point (for
example RS) we know about real  customer (inbound) and supplier
(outbound).

Currently, RS assumes MS_N as outbound supplier because real supplier
is behind MS_N. I hope you have got my point.


Thanks


On Sun, May 30, 2010 at 11:45 AM, Vitalii Colosov <vetali100 at gmail.com>wrote:

> Could you please provide an example of what exactly do you think is a
> problem?
> Because now I don't have any problem with this configuration and I am
> wondering what is the exact scenario where we cannot use such configuration.
>
> Thank you,
> Vitalie
>
> 2010/5/30 Saeed Ahmed <saeedahmad1981 at gmail.com>
>
> Another point, with following script:
>>
>> media_server="my_media_X.mydomain.com <http://my_media_x.mydomain.com/>";
>> --to be determined by routing logic
>> forwarding_session = "sofia/external/"..called_number.."@"..media_server;
>> session:setVariable("bypass_media_after_bridge", "true");
>> session:setVariable("hangup_after_bridge", "true");
>> session:execute("bridge",forwarding_session);
>>
>> we can bill the customer (inbound) but supplier (outbound) is behind
>> media_server, so i think it would be a problem to manage ,supplier side for
>> billing and statistics, on signalling server.
>>
>> On Sat, May 29, 2010 at 10:55 PM, Vitalii Colosov <vetali100 at gmail.com>wrote:
>>
>>> In our implementation only Media servers generate CDRs. No Media session
>>> - no CDR (at least we need exactly this behavior).
>>>
>>> However if you need, you can generate CDRs only on Signalling servers I
>>> beleive.
>>>
>>> Regards,
>>> Vitalie
>>>
>>> 2010/5/29 Saeed Ahmed <saeedahmad1981 at gmail.com>
>>>
>>> If i understood correctly, Vitalie solutions is still workable,
>>>> (although what Code mentioned, would be ideal), because from customer side
>>>> its normal to provide multiple IPs or in most cases a whole subnet range,
>>>> and call can come any IP from theatrange, a good example is Arbinet.
>>>>
>>>> Commercial SBCs like nextone support it.
>>>>
>>>> Vitalie, i've a concern that in your solution how would we deal with
>>>> cdrs?
>>>>
>>>> Thanks
>>>>
>>>> On Fri, May 28, 2010 at 1:31 PM, David Ponzone <david.ponzone at gmail.com
>>>> > wrote:
>>>>
>>>>> Code,
>>>>>
>>>>> you're totally right.
>>>>> In this model (FS), the media server will also be in the SIP Path.
>>>>> That's why I answered in the first place that this was not achievable
>>>>> with FS, because your idea was more a Kamaillo/RTPProxy setup, where the
>>>>> mediaserver only does RTP with the endpoints, and is not in the SIP path at
>>>>> all:
>>>>>
>>>>> inbound <--------SIP------ SIP Server/Proxy ------------SIP------->
>>>>> Carrier
>>>>>                                                   |
>>>>>              <---------RTP------ MediaServer--------RTP--------------->
>>>>>
>>>>>
>>>>> Verizon Business (in Europe at least) has a such infrastrucutre:
>>>>> OpenSER for the SIP part, and Nortel GWs for the RTP.
>>>>> This way, they just give me the IPs of their OpenSER servers, and they
>>>>> can deploy as many media servers as they need without telling us (of course,
>>>>> we dont filter that).
>>>>>
>>>>> I don't know how this is implemented in Kamaillo/OpenSER but perhaps,
>>>>> for a nice bounty, that would be something possible in FS.
>>>>>
>>>>> David Ponzone  Direction Technique
>>>>>  email: david.ponzone at ipeva.fr
>>>>> tel:      01 74 03 18 97
>>>>> gsm:   06 66 98 76 34
>>>>>
>>>>> Service Client IPeva
>>>>> tel:      0811 46 26 26
>>>>> www.ipeva.fr  -   www.ipeva-studio.com
>>>>>
>>>>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
>>>>> ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>>>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>>>>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>>>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>>>>> l'exp?diteur.*
>>>>> *
>>>>> *
>>>>>
>>>>>
>>>>>
>>>>> Le 28/05/2010 ? 05:34, Code Ghar a ?crit :
>>>>>
>>>>> Hi Vitalie
>>>>>
>>>>> Thanks for providing the link and details. If I understood correctly,
>>>>> the chain of signaling would be Inbound -> FSSIP -> FSRTP -> Outbound (using
>>>>> names and terms in my original question), while the chain of media would be
>>>>> Inbound -> FSRTP -> Outbound. This way we can have multiple servers handling
>>>>> media and minimal servers handling signaling.
>>>>>
>>>>> Let me clarify a little more my motivation for asking this question in
>>>>> the first place. I work with telecom carriers on a daily basis and have seen
>>>>> many different architectures. The first biggest problem is how to load
>>>>> balance SIP traffic when you are receiving calls, if one server is not
>>>>> enough. The second biggest problem is handling all RTP, including
>>>>> transcoding. With this architecture, one or two IPs for signaling can be
>>>>> handled by most carriers. So you can beef up your hardware for signaling and
>>>>> depend less on your carrier's ability to load balance traffic for you. If
>>>>> they can do round-robin or failover for two IPs, you are golden. And then
>>>>> you can deploy multiple nodes to handle all RTP duties, without having to
>>>>> concern your carrier about multiple servers and IPs. But there's one thing
>>>>> still missing. Your outbound carrier still needs to allow traffic from
>>>>> multiple IPs because now they are dealing with FSRTP instead of FSSIP.
>>>>>
>>>>> Is there such a solution possible using FS that all signaling, for both
>>>>> inbound and outbound carriers, can be handled by a couple of FSSIP nodes
>>>>> (depending on the amount of traffic, maybe a few more) while delegating
>>>>> media responsibilities to multiple FSRTP nodes? In this situation, SIP IP is
>>>>> always, say 10.10.10.1 or 10.10.10.2, but the SDP may use media IPs
>>>>> 10.10.10.3, 10.10.10.4, 10.10.10.5, and so on. Almost all carriers I have
>>>>> seen concern themselves only with which SIP IPs they should allow and don't
>>>>> care how many or which media IPs they have to deal with. Any ideas on
>>>>> minimizing signaling IPs while adding more media IPs as required?
>>>>>
>>>>> I have seen re-invite being used in production where you can just let
>>>>> your inbound and outbound handle media between them on their own without it
>>>>> going through your network. But there are circumstances where people might
>>>>> need to pass media through their own network, chiefly to perform transcoding
>>>>> and also to handle other interop issues. It is because of this use case,
>>>>> combined with the need for minimizing signaling IPs, that the original
>>>>> question was asked.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Thu, May 27, 2010 at 1:15 AM, Vitalii Colosov <vetali100 at gmail.com>wrote:
>>>>>
>>>>>> Hi Code,
>>>>>>
>>>>>> I have working example doing exactly what you've described.
>>>>>> One signalling FS bridges incoming call to a set of media servers
>>>>>> (depending on ip, but you can implement any routing logic including round
>>>>>> robin) and then transfers media stream after bridging to that media server.
>>>>>>
>>>>>> You can achieve this on signalling FS by creating a Lua script that
>>>>>> contains the following lines:
>>>>>>
>>>>>> media_server="my_media_X.mydomain.com"; --to be determined by routing
>>>>>> logic
>>>>>> forwarding_session =
>>>>>> "sofia/external/"..called_number.."@"..media_server;
>>>>>> session:setVariable("bypass_media_after_bridge", "true");
>>>>>> session:setVariable("hangup_after_bridge", "true");
>>>>>> session:execute("bridge",forwarding_session);
>>>>>>
>>>>>> The call will arrive to the selected media server successfully and
>>>>>> media stream will start bypassing signalling FS after bridge.
>>>>>>
>>>>>> You can read the following thread, it describes how you can setup such
>>>>>> configuration.
>>>>>>
>>>>>> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-March/055231.html
>>>>>>
>>>>>> I think it will fit your needs.
>>>>>>
>>>>>> Regards,
>>>>>> Vitalie
>>>>>>
>>>>>>
>>>>>> 2010/5/27 Code Ghar <codeghar at gmail.com>
>>>>>>
>>>>>>> Is it possible -- and are there any case studies, practical
>>>>>>> experience, etc -- on deploying FreeSWITCH (FS) in this architecture: one
>>>>>>> server (FSSIP) handles SIP signaling only, and multiple servers (FSRTP1,
>>>>>>> FSRTP2, ..., FSRTPn) handle all media responsibilities? So when a call comes
>>>>>>> in, the SDP contains IP of, say FSRTP1, as media handler. For this to work,
>>>>>>> FSSIP would request FSRTPx for media resources for each new call and add its
>>>>>>> IP and port in SDP. The media servers/gateways would play IVR, etc.; collect
>>>>>>> DTMF and forward as appropriate to FSSIP; perform transcoding; etc.; all
>>>>>>> while FSSIP only deals with signaling. This way multiple servers could be
>>>>>>> deployed to handle media responsibilities and only a handful would be
>>>>>>> required for signaling. In future if there's a greater need for transcoding,
>>>>>>> etc. all you need to do is deploy a media server and not have to add servers
>>>>>>> for signaling.
>>>>>>>
>>>>>>> This idea came to me because I have come across two proprietary
>>>>>>> applications that do it this way. They have a SIP component and a media
>>>>>>> component. You can run both on the same physical machine or you can separate
>>>>>>> them out into multiple machines.
>>>>>>>
>>>>>>> Another way for this could be to integrate FS as a media component to
>>>>>>> another application's SIP component. A mix-and-match, so to speak.
>>>>>>>
>>>>>>> On the flip side, deploy FS as a SIP server and use media
>>>>>>> capabilities of some other hardware or software application. For example, FS
>>>>>>> handles signaling and use dedicated hardware for media. A good example of
>>>>>>> this is illustrated (somewhat) by an image on Sangoma's website:
>>>>>>> http://www.sangoma.com/assets/images/content/transcoding_diagram.jpg.
>>>>>>> Look at the "pooled transcoding".
>>>>>>>
>>>>>>> Is FS even designed to be this modular? If so, how can the
>>>>>>> aforementioned scenario(s) be achieved?
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:
>>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From kees at mroffice.org  Sun May 30 06:17:23 2010
From: kees at mroffice.org (Kees Varekamp)
Date: Mon, 31 May 2010 01:17:23 +1200
Subject: [Freeswitch-users] recording to ogg vorbis
In-Reply-To: <4C02296F.1080407@gmail.com>
References: <AANLkTine9yRZCLFMepJqQozZhdXJ9NXI6QYmaLsHWRId@mail.gmail.com>
	<D13190E0-926B-4989-9FC5-CEABD4124077@freeswitch.org>
	<AANLkTim4OGNAtPR0v23CSZaHIdklgDBDM3HCDopPQrSf@mail.gmail.com>
	<AE9F2D1E-00B0-4066-8FFC-78BBDF5ABCC4@freeswitch.org>
	<AANLkTinIifgtI88Y7VdL_orE9d1xvKBFyBSWUGZ17O0C@mail.gmail.com>
	<4C02296F.1080407@gmail.com>
Message-ID: <AANLkTinpYUPCYcbcVB1inysByoY_Y_MBNWCnYMa56T1z@mail.gmail.com>

Thanks.

I think I got it fixed:

1) libsndfile needs updating to 1.0.21
2) libsndfile needs libogg, libvorbis, libflac and pkg-config (otherwise it
will happily compile, but without ogg support)
3) mod_sndfile needs an extra else if after line 150:
} else if (!strcmp(ext, "oga")) {
context->sfinfo.format = SF_FORMAT_OGG | SF_FORMAT_VORBIS;
context->sfinfo.channels = 1;
context->sfinfo.samplerate = 8000;
}

4) make && make install mod_sndfile
5) ...
6) Profit!

I can record *.oga files now :-))) Very cool.


On Sun, May 30, 2010 at 21:01, Tamas <jalsot at gmail.com> wrote:

>  Hello,
>
> maybe upgrading libsndfile could do the trick as new version supports
> OGG/Vorbis.
> (http://www.mega-nerd.com/libsndfile/).
>
> Regards,
>     Tamas
>
> 2010-05-30 09:24 keltez?ssel, Kees Varekamp ?rta:
>
> I am keen but my c skills are not top notch. But I'll give it a go.
>
>  tnx,
> Kees
>
> On Sun, May 30, 2010 at 18:52, Brian West <brian at freeswitch.org> wrote:
>
>> you could write a format module to allow that.
>>
>> Sent from my iPad
>>
>> On May 30, 2010, at 1:34 AM, Kees Varekamp <kees at mroffice.org> wrote:
>>
>> > OK tnx :-)
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From brian at freeswitch.org  Sun May 30 07:13:38 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 30 May 2010 09:13:38 -0500
Subject: [Freeswitch-users] recording to ogg vorbis
In-Reply-To: <AANLkTinpYUPCYcbcVB1inysByoY_Y_MBNWCnYMa56T1z@mail.gmail.com>
References: <AANLkTine9yRZCLFMepJqQozZhdXJ9NXI6QYmaLsHWRId@mail.gmail.com>
	<D13190E0-926B-4989-9FC5-CEABD4124077@freeswitch.org>
	<AANLkTim4OGNAtPR0v23CSZaHIdklgDBDM3HCDopPQrSf@mail.gmail.com>
	<AE9F2D1E-00B0-4066-8FFC-78BBDF5ABCC4@freeswitch.org>
	<AANLkTinIifgtI88Y7VdL_orE9d1xvKBFyBSWUGZ17O0C@mail.gmail.com>
	<4C02296F.1080407@gmail.com>
	<AANLkTinpYUPCYcbcVB1inysByoY_Y_MBNWCnYMa56T1z@mail.gmail.com>
Message-ID: <3B3D4A67-802C-42E0-BCC9-D61F8D366517@freeswitch.org>

See how easy things can be sometimes.. can you post a patch on jira?

/b

On May 30, 2010, at 8:17 AM, Kees Varekamp wrote:

> Thanks.
> 
> I think I got it fixed:
> 
> 1) libsndfile needs updating to 1.0.21
> 2) libsndfile needs libogg, libvorbis, libflac and pkg-config (otherwise it will happily compile, but without ogg support)
> 3) mod_sndfile needs an extra else if after line 150:
> 	} else if (!strcmp(ext, "oga")) {
> 		context->sfinfo.format = SF_FORMAT_OGG | SF_FORMAT_VORBIS;
> 		context->sfinfo.channels = 1;
> 		context->sfinfo.samplerate = 8000;
> 	}
> 
> 4) make && make install mod_sndfile 
> 5) ...
> 6) Profit!
> 
> I can record *.oga files now :-))) Very cool.
> 



From brian at freeswitch.org  Sun May 30 07:21:37 2010
From: brian at freeswitch.org (Brian West)
Date: Sun, 30 May 2010 09:21:37 -0500
Subject: [Freeswitch-users] recording to ogg vorbis
In-Reply-To: <AANLkTinpYUPCYcbcVB1inysByoY_Y_MBNWCnYMa56T1z@mail.gmail.com>
References: <AANLkTine9yRZCLFMepJqQozZhdXJ9NXI6QYmaLsHWRId@mail.gmail.com>
	<D13190E0-926B-4989-9FC5-CEABD4124077@freeswitch.org>
	<AANLkTim4OGNAtPR0v23CSZaHIdklgDBDM3HCDopPQrSf@mail.gmail.com>
	<AE9F2D1E-00B0-4066-8FFC-78BBDF5ABCC4@freeswitch.org>
	<AANLkTinIifgtI88Y7VdL_orE9d1xvKBFyBSWUGZ17O0C@mail.gmail.com>
	<4C02296F.1080407@gmail.com>
	<AANLkTinpYUPCYcbcVB1inysByoY_Y_MBNWCnYMa56T1z@mail.gmail.com>
Message-ID: <BDBA23BE-F4EB-4C43-8850-0F6BF3D9A88C@freeswitch.org>

Ogg can record native sample rates I would say context->sfinof.samplerate= handle->samplerate;  

This would mean that it would record at the channels native sample rate and no resample would be required to record 8,12,16,24,32 or 48kHz channels.

Could flac support be added also?

/b

On May 30, 2010, at 8:17 AM, Kees Varekamp wrote:

> Thanks.
> 
> I think I got it fixed:
> 
> 1) libsndfile needs updating to 1.0.21
> 2) libsndfile needs libogg, libvorbis, libflac and pkg-config (otherwise it will happily compile, but without ogg support)
> 3) mod_sndfile needs an extra else if after line 150:
> 	} else if (!strcmp(ext, "oga")) {
> 		context->sfinfo.format = SF_FORMAT_OGG | SF_FORMAT_VORBIS;
> 		context->sfinfo.channels = 1;
> 		context->sfinfo.samplerate = 8000;
> 	}
> 
> 4) make && make install mod_sndfile 
> 5) ...
> 6) Profit!



From david.ponzone at gmail.com  Sun May 30 11:59:09 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Sun, 30 May 2010 20:59:09 +0200
Subject: [Freeswitch-users] FS as Media Gateway Only
In-Reply-To: <AANLkTim5o5dc84N4Ag9PenNKvVPfrfr2cQWIQmjVhWuT@mail.gmail.com>
References: <AANLkTimgKRtL06xmyjXUPnpNIjlYkHC6BytzmTqKmVm7@mail.gmail.com>
	<AANLkTikoxsXFuYBxsXXtB26Y9_8f1I1wGmE-u-I_l9PP@mail.gmail.com>
	<AANLkTikgol1TigK9G9vvk90ogtjqeJ3nwPXqdFUXcV5W@mail.gmail.com>
	<BE75B7EE-B77D-4710-899E-AE9E771A28F7@gmail.com>
	<AANLkTimQUri22GlLok8z8LGSFRmufCIXdWGdA4NStbfI@mail.gmail.com>
	<AANLkTimI28Pi_oRNRkgNtT4dT9_cLTPJ9a2XVkgGFzAW@mail.gmail.com>
	<AANLkTimBbGprWu53lQqWhaTmLRdh25l-yCmYbbT1MXBq@mail.gmail.com>
	<AANLkTimSpVOYVB2BiKZWVr3dNrjy0ufQv7YsqAAbdRxS@mail.gmail.com>
	<AANLkTim5o5dc84N4Ag9PenNKvVPfrfr2cQWIQmjVhWuT@mail.gmail.com>
Message-ID: <DDABF8BB-CB0B-4AE1-9504-8F87CD894D59@gmail.com>

Well, I think I have a way to solve that.

On RS, set a custom SIP header before bridging the call:
X-Route: Supplier-X

And then in the MS dialplan, you just have to route to the right  
gateway according to X-Route.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 30/05/2010 ? 13:21, Saeed Ahmed a ?crit :

> Hi,
>
> Actually i am trying to setup the environment to test it real time,  
> but theoretically it looks like that:
>
>                --> MS_1 ------> Supplier (outbound)
> Inbound ---->  RS  --> MS_2 --> Supplier (outbound)
>                --> MS_N ------> Supplier (outbound)
> So what i would like to do, when call hits RS, it will look in  
> database (using curl or lua etc..) for end supplier (outbound) and  
> also selects MS_N (to deal RTP media), so that on a central point  
> (for example RS) we know about real  customer (inbound) and supplier  
> (outbound).
>
> Currently, RS assumes MS_N as outbound supplier because real  
> supplier is behind MS_N. I hope you have got my point.
>
> Thanks
>
> On Sun, May 30, 2010 at 11:45 AM, Vitalii Colosov  
> <vetali100 at gmail.com> wrote:
> Could you please provide an example of what exactly do you think is  
> a problem?
> Because now I don't have any problem with this configuration and I  
> am wondering what is the exact scenario where we cannot use such  
> configuration.
>
> Thank you,
> Vitalie
>
> 2010/5/30 Saeed Ahmed <saeedahmad1981 at gmail.com>
>
> Another point, with following script:
>
> media_server="my_media_X.mydomain.com"; --to be determined by  
> routing logic
> forwarding_session = "sofia/ 
> external/"..called_number.."@"..media_server;
> session:setVariable("bypass_media_after_bridge", "true");
> session:setVariable("hangup_after_bridge", "true");
> session:execute("bridge",forwarding_session);
>
> we can bill the customer (inbound) but supplier (outbound) is behind  
> media_server, so i think it would be a problem to manage ,supplier  
> side for billing and statistics, on signalling server.
>
> On Sat, May 29, 2010 at 10:55 PM, Vitalii Colosov  
> <vetali100 at gmail.com> wrote:
> In our implementation only Media servers generate CDRs. No Media  
> session - no CDR (at least we need exactly this behavior).
>
> However if you need, you can generate CDRs only on Signalling  
> servers I beleive.
>
> Regards,
> Vitalie
>
> 2010/5/29 Saeed Ahmed <saeedahmad1981 at gmail.com>
>
> If i understood correctly, Vitalie solutions is still workable,  
> (although what Code mentioned, would be ideal), because from  
> customer side its normal to provide multiple IPs or in most cases a  
> whole subnet range, and call can come any IP from theatrange, a good  
> example is Arbinet.
>
> Commercial SBCs like nextone support it.
>
> Vitalie, i've a concern that in your solution how would we deal with  
> cdrs?
>
> Thanks
>
> On Fri, May 28, 2010 at 1:31 PM, David Ponzone <david.ponzone at gmail.com 
> > wrote:
> Code,
>
> you're totally right.
> In this model (FS), the media server will also be in the SIP Path.
> That's why I answered in the first place that this was not  
> achievable with FS, because your idea was more a Kamaillo/RTPProxy  
> setup, where the mediaserver only does RTP with the endpoints, and  
> is not in the SIP path at all:
>
> inbound <--------SIP------ SIP Server/Proxy ------------SIP------->  
> Carrier
>                                                   |
>              <---------RTP------ MediaServer-------- 
> RTP--------------->
>
>
> Verizon Business (in Europe at least) has a such infrastrucutre:  
> OpenSER for the SIP part, and Nortel GWs for the RTP.
> This way, they just give me the IPs of their OpenSER servers, and  
> they can deploy as many media servers as they need without telling  
> us (of course, we dont filter that).
>
> I don't know how this is implemented in Kamaillo/OpenSER but  
> perhaps, for a nice bounty, that would be something possible in FS.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> Ce message et toutes les pi?ces jointes sont confidentiels et  
> ?tablis ? l'intention exclusive de ses destinataires. Toute  
> utilisation ou diffusion non autoris?e est interdite. Tout message  
> ?lectronique est susceptible d'alt?ration. IPeva d?cline toute  
> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou  
> falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le  
> d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
>
> Le 28/05/2010 ? 05:34, Code Ghar a ?crit :
>
>> Hi Vitalie
>>
>> Thanks for providing the link and details. If I understood  
>> correctly, the chain of signaling would be Inbound -> FSSIP ->  
>> FSRTP -> Outbound (using names and terms in my original question),  
>> while the chain of media would be Inbound -> FSRTP -> Outbound.  
>> This way we can have multiple servers handling media and minimal  
>> servers handling signaling.
>>
>> Let me clarify a little more my motivation for asking this question  
>> in the first place. I work with telecom carriers on a daily basis  
>> and have seen many different architectures. The first biggest  
>> problem is how to load balance SIP traffic when you are receiving  
>> calls, if one server is not enough. The second biggest problem is  
>> handling all RTP, including transcoding. With this architecture,  
>> one or two IPs for signaling can be handled by most carriers. So  
>> you can beef up your hardware for signaling and depend less on your  
>> carrier's ability to load balance traffic for you. If they can do  
>> round-robin or failover for two IPs, you are golden. And then you  
>> can deploy multiple nodes to handle all RTP duties, without having  
>> to concern your carrier about multiple servers and IPs. But there's  
>> one thing still missing. Your outbound carrier still needs to allow  
>> traffic from multiple IPs because now they are dealing with FSRTP  
>> instead of FSSIP.
>>
>> Is there such a solution possible using FS that all signaling, for  
>> both inbound and outbound carriers, can be handled by a couple of  
>> FSSIP nodes (depending on the amount of traffic, maybe a few more)  
>> while delegating media responsibilities to multiple FSRTP nodes? In  
>> this situation, SIP IP is always, say 10.10.10.1 or 10.10.10.2, but  
>> the SDP may use media IPs 10.10.10.3, 10.10.10.4, 10.10.10.5, and  
>> so on. Almost all carriers I have seen concern themselves only with  
>> which SIP IPs they should allow and don't care how many or which  
>> media IPs they have to deal with. Any ideas on minimizing signaling  
>> IPs while adding more media IPs as required?
>>
>> I have seen re-invite being used in production where you can just  
>> let your inbound and outbound handle media between them on their  
>> own without it going through your network. But there are  
>> circumstances where people might need to pass media through their  
>> own network, chiefly to perform transcoding and also to handle  
>> other interop issues. It is because of this use case, combined with  
>> the need for minimizing signaling IPs, that the original question  
>> was asked.
>>
>>
>>
>>
>> On Thu, May 27, 2010 at 1:15 AM, Vitalii Colosov  
>> <vetali100 at gmail.com> wrote:
>> Hi Code,
>>
>> I have working example doing exactly what you've described.
>> One signalling FS bridges incoming call to a set of media servers  
>> (depending on ip, but you can implement any routing logic including  
>> round robin) and then transfers media stream after bridging to that  
>> media server.
>>
>> You can achieve this on signalling FS by creating a Lua script that  
>> contains the following lines:
>>
>> media_server="my_media_X.mydomain.com"; --to be determined by  
>> routing logic
>> forwarding_session = "sofia/ 
>> external/"..called_number.."@"..media_server;
>> session:setVariable("bypass_media_after_bridge", "true");
>> session:setVariable("hangup_after_bridge", "true");
>> session:execute("bridge",forwarding_session);
>>
>> The call will arrive to the selected media server successfully and  
>> media stream will start bypassing signalling FS after bridge.
>>
>> You can read the following thread, it describes how you can setup  
>> such configuration.
>> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-March/055231.html
>>
>> I think it will fit your needs.
>>
>> Regards,
>> Vitalie
>>
>>
>> 2010/5/27 Code Ghar <codeghar at gmail.com>
>> Is it possible -- and are there any case studies, practical  
>> experience, etc -- on deploying FreeSWITCH (FS) in this  
>> architecture: one server (FSSIP) handles SIP signaling only, and  
>> multiple servers (FSRTP1, FSRTP2, ..., FSRTPn) handle all media  
>> responsibilities? So when a call comes in, the SDP contains IP of,  
>> say FSRTP1, as media handler. For this to work, FSSIP would request  
>> FSRTPx for media resources for each new call and add its IP and  
>> port in SDP. The media servers/gateways would play IVR, etc.;  
>> collect DTMF and forward as appropriate to FSSIP; perform  
>> transcoding; etc.; all while FSSIP only deals with signaling. This  
>> way multiple servers could be deployed to handle media  
>> responsibilities and only a handful would be required for  
>> signaling. In future if there's a greater need for transcoding,  
>> etc. all you need to do is deploy a media server and not have to  
>> add servers for signaling.
>>
>> This idea came to me because I have come across two proprietary  
>> applications that do it this way. They have a SIP component and a  
>> media component. You can run both on the same physical machine or  
>> you can separate them out into multiple machines.
>>
>> Another way for this could be to integrate FS as a media component  
>> to another application's SIP component. A mix-and-match, so to speak.
>>
>> On the flip side, deploy FS as a SIP server and use media  
>> capabilities of some other hardware or software application. For  
>> example, FS handles signaling and use dedicated hardware for media.  
>> A good example of this is illustrated (somewhat) by an image on  
>> Sangoma's website: http://www.sangoma.com/assets/images/content/transcoding_diagram.jpg 
>> . Look at the "pooled transcoding".
>>
>> Is FS even designed to be this modular? If so, how can the  
>> aforementioned scenario(s) be achieved?
>>
>> _______________________________________________
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From mitch.capper at gmail.com  Sun May 30 12:09:33 2010
From: mitch.capper at gmail.com (Mitch Capper)
Date: Sun, 30 May 2010 15:09:33 -0400
Subject: [Freeswitch-users] Portaudio: Call For Input
Message-ID: <AANLkTimrhkeQRSnpj-9T6_5rCTmMcfukU3fpZBXvpakp@mail.gmail.com>

Recently I have started attempting to replace an existing sip client with
freeswitch.  In doing so I have started looking more towards portaudio.   My
goal is to add some robustness to portaudio to make it better suited to
being used by sip clients.    Some of the features I have currently worked
to implement:

*Event generation for call holding and resuming
*Ability to switch input/output devices during calls with live audio going
on (useful for say speakerphone support)
*Ability to switch both the indev and outdev at once rather than separately
(time savings)
*Ability to keep the audio stream initialized rather than each time a call
is made/ device is rang (time savings)
*Only init codecs during load / config reload rather than on audio stream
init (time savings)
*Ability to call play (for sounds) with active calls going on

Most of the improvements so far have been to make portaudio a bit faster at
doing things and less restricted than previously coded.  Most changes have
been made through additional configuration variables so out of the box
portaudio will not function any differently than currently.   My request for
input is on next steps and warnings.

Right now portaudio keeps track of two streams, the call audio stream and
the ring audio stream (and both are inited on demand only accept with my
always active stream change).

I was thinking of taking the always active streams to a slightly higher
level:
  I am thinking about moving to have it keep track of an arbitrary number of
streams,  a linked list of streams it is keeping track of with no more than
one stream per input/output pair.    This would allow for initing a device
prior to use to allow for near instant use of that stream.  This would
remove the small delay that still exists (under 1 second currently I would
estimate) when say switching on the speaker phone.  Or allow for very quick
playing of audio on a specific device.
Aside from the speed increases in stream switching this would allow for you
to play audio on a device that isnt currently the primary audio device.
While I do not plan to take it this far currently, it would actually allow
for a much easier time of handling multiple calls on different devices at
the same time.

The other way to go is to look towards trying to speed up initing streams
more, there are some yields in the code that look like safety things that
may be possible to remove without negative affects.


Also I looked at the last merge of portaudio from upstream into trunk (end
of 2007) and the changes that were made back then.  The good news is there
were not actually a lot of base code changes and most of them have actually
been merged into upstream now so updating to the latest may not be a very
hard thing to do.   This is in part to see if it helps with the audio
quality issues that people of portaudio seem to report as it certainly is
not something you want in your client.


If anyone has any input into the current state of portaudio or purposed
changes please let me know.   In addition if anyone has input into getting
better call quality out of PA that would be extremely advantageous, as other
than updating to trunk I doubt my changes will result in much of a quality
improvement.   I am specifically also interested if anyone knows of any of
the reasons some of the safeguards that are there are in place, some of
which I don't see a technical reason for them to be there unless there is
funnyness in libportaudio itself  (which certainly could be and may be
partially resolved with updating to the changes from the last 3 years).   If
anyone knows any stress testing etc that presented issues for PA previously
that would be helpful.  I am working with PA in windows, do have a linux box
I will do some limited testing on, but overall if you just can test various
changes let me know as that will be helpful too as I believe removing some
of the safety steps may not present as an issue right away. I could also add
a "faster" option to portaudio config that would result in the optimizations
rather than just removing them completely.



~Mitch
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From bobc at devassert.com  Sun May 30 14:55:38 2010
From: bobc at devassert.com (Bob Coleman)
Date: Mon, 31 May 2010 09:55:38 +1200
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
In-Reply-To: <4BFEE96A.2010003@gmail.com>
References: <490aedfdd7c8ac680bbe0d7f68c786de@mail.gmail.com>
	<4BFBE6C2.5000703@gmail.com>
	<AANLkTilsHn9znajzF0Jj-QDu4je9QW7CeymuBYE0SyKa@mail.gmail.com>
	<4BFBEF7C.9020107@gmail.com>
	<9250A9FF05DB4B879DEF6ECE8E3FFAC1@dell9400>
	<AANLkTilU2OwgH0Qk_Qoc3R7ErPO_Z_VRhePusetkRkeW@mail.gmail.com>
	<AANLkTikXWvEB_C3jqOWLHaVcns9-U5Hok-5MqgE_uzzZ@mail.gmail.com>
	<4BFC4A64.5090208@gmail.com>
	<AANLkTinFtVXa7CPou-ShPYIXHq0AktokhptGSEFKBkge@mail.gmail.com>
	<D311BDCB06324D0DB77898FC6FF4B0EB@dell9400>
	<AANLkTilAMe1lRKK4R6Nmr29FzfI-pP3wRa7Foabmd8Dp@mail.gmail.com>
	<4BFD4562.5030804@gmail.com>
	<AANLkTikGQlZ4N3zfYv18COvbubEi-8LSOZxnAy1KC5sP@mail.gmail.com>
	<4BFEE96A.2010003@gmail.com>
Message-ID: <AANLkTikSgh4xN9kb3Lha5dNlacgTA9DqrkHqU3qCWZUT@mail.gmail.com>

I have put up an example for the outbound eventsocket in c#, it is
real basic but gives you one approach to using this method. This has
been coupled together by looking at other samples.

There are two files, one is the dialplan ( 810_sample_ivr.xml ) which
should be placed in the conf/dialplan/default folder
The other file is a zip including a library for talking to the event
socket and a simple test listener.

http://www.devassert.com/apps/freeswitch/810_sample_ivr.xml
http://www.devassert.com/apps/freeswitch/freeswitch.es.net.zip

The password on the zip is cluecon

To try the test, have the dialplan in place and run freeswitch,
connect to an extension with a soft phone, eg 1001, dial 810 and
listen.

Assuming you have freeswitch callie prompts installed
sounds\en\us\callie\conference\8000 all will be good.

Sample is in Visual Studio 2008 and uses a threading library written
by another party.

If you have any questions just ask.

On Fri, May 28, 2010 at 9:51 AM, Phone <phone.bytes at gmail.com> wrote:
> Thanks,
>
> Examples and feedback are most helpful!
>
> Bob Coleman wrote:
>> Hi,
>>
>> Will incorporate some threading into the example I am working on for you.
>>
>> The outbound event socket method is very similar to the dialogic
>> environment, I know how you felt though, but by starting small(like
>> just even answering a call) gets you moving pretty quick. Freeswitch
>> is lots of fun to work with, and the guys on here are very
>> supportive!!
>>
>> Bob
>>
>> On Thu, May 27, 2010 at 3:59 AM, Phone <phone.bytes at gmail.com> wrote:
>>
>>> Thanks to all for the most helpful feedback. ?Sharing your approaches
>>> and experiences are a big help. ?I look forward to the upcoming code
>>> samples.
>>>
>>> I was coming from a windows/dialogic environment where I used a library
>>> that allowed me to work on a little higher level. ?For example, I had a
>>> call to "play a file" that took a parameter of whether or not to allow a
>>> dtmf to interrupt. ?There was also a call to "ReadDtmfs" that took
>>> parameters to specify the number of Dtmf's to read, how long to wait for
>>> them, and what terminating character to use. ?I guess that you could
>>> write some scripts or compiled code with these same types of functions
>>> to simplify some of these routine tasks with reusable code?
>>>
>>> Also, the library handled the threading and scheduling with the OS. ?I
>>> am still unclear on handling the events. ?I guess you have a big loop
>>> reading events and then acting on them using the uuid to determine which
>>> call it is and how to deal with the next step of the call? ?Any feedback
>>> on this part of the project?
>>>
>>> Again, Thanks!
>>>
>>> Bob Coleman wrote:
>>>
>>>> Ah sorry, I started with the esl to get an understanding then wrote my
>>>> own socket library(was actually very easy to do), when I mean docs I
>>>> mean the event socket docs. I still think of it as the esl, my
>>>> mistake.
>>>>
>>>> http://wiki.freeswitch.org/wiki/Event_Socket
>>>>
>>>> I started with a codeplex project, that had been abandoned, and then
>>>> once I understood the structure of the event socket language, was able
>>>> to rewrite it to better handle what we were doing.
>>>>
>>>> I also married it up to an old gotdotnet asterisk fast agi project,
>>>> once again abandoned, to allow for the use of asterisk as well, but in
>>>> the end freeswitch won because we could use just one platform.
>>>>
>>>> I am busy writing a small sample app at the moment to demonstrate a
>>>> problem I am trying to solve. Can release that code once sorted. Will
>>>> be in a week or so. Am intending it as a quick way of testing event
>>>> sockets, and trying various commands etc. before commiting to coding
>>>> something.
>>>>
>>>> Bob
>>>>
>>>> On Wed, May 26, 2010 at 2:26 PM, Jan Berger <jan.berger at video24.no> wrote:
>>>>
>>>>
>>>>> Hi,
>>>>>
>>>>> Do you have some sample code you could share + what docs did you look at?
>>>>>
>>>>> I would like to write and test some C# using ESL for my own work.
>>>>>
>>>>> Jan
>>>>>
>>>>> -----Original Message-----
>>>>> From: freeswitch-users-bounces at lists.freeswitch.org
>>>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bob
>>>>> Coleman
>>>>> Sent: 26. mai 2010 04:07
>>>>> To: freeswitch-users at lists.freeswitch.org
>>>>> Subject: Re: [Freeswitch-users] Questions on Building an application for
>>>>> FreeSWITCH
>>>>>
>>>>> We used c# as the rest of our systems are windows based. The language
>>>>> doesnt matter too much, as long as you know where you are headed, what
>>>>> performance you require, and what platform you are going to be using.
>>>>>
>>>>> Found the ESL so much easier than the dialogic c library we were using.
>>>>>
>>>>> The docs for the esl are easy to understand, the thing I couldnt get
>>>>> my head around initially was the dialing out, with the dialogic you
>>>>> are in the middle when you dial, ie already on the channel, but with
>>>>> freeswitch you are kind of the third party when you dial, the channel
>>>>> being created by the dialing and handing it off to be worked on. We
>>>>> make the call via an inbound event socket and hand it off to an
>>>>> outbound event socket application via the dialplan.
>>>>>
>>>>> On Wed, May 26, 2010 at 10:08 AM, Phone <phone.bytes at gmail.com> wrote:
>>>>>
>>>>>
>>>>>> Thanks for the info. ?What language did you use?
>>>>>>
>>>>>> Bob Coleman wrote:
>>>>>>
>>>>>>
>>>>>>> Have just recently completed a project to convert an old windows
>>>>>>> dialogic application(vb6) to FreeSWITCH, would reccommend using the
>>>>>>> ESL, was able to map the old dialogic calls to the ESL calls pretty
>>>>>>> easily. We used a mixture of inbound and outbound sockets, as we have
>>>>>>> people dialing us, not just dialing out etc.
>>>>>>>
>>>>>>> With the dialogic you open a port and make the call and handle the
>>>>>>> dtmf, with freeswitch you create a socket connection to FreeSWITCH to
>>>>>>> dial the number and then hand it off to an extension for processing
>>>>>>> the dtmf(that is one approach any way)
>>>>>>>
>>>>>>> Bob
>>>>>>>
>>>>>>> On Wed, May 26, 2010 at 6:45 AM, Michael Collins <msc at freeswitch.org>
>>>>>>>
>>>>>>>
>>>>> wrote:
>>>>>
>>>>>
>>>>>>>> On Tue, May 25, 2010 at 11:38 AM, Jan Berger <jan.berger at video24.no>
>>>>>>>>
>>>>>>>>
>>>>> wrote:
>>>>>
>>>>>
>>>>>>>>> Actually - before you get "to smart" - may I suggest that you start
>>>>>>>>> writing
>>>>>>>>> - or improving - the getting started sections of the doc. Address the
>>>>>>>>> areas
>>>>>>>>> where you struggle and let others benefit from your work.
>>>>>>>>>
>>>>>>>>> I have been through similar issues myself - FS is one of the easier
>>>>>>>>> projects
>>>>>>>>> to work with once you get under the hood, but you basically need to
>>>>>>>>>
>>>>>>>>>
>>>>> evolve
>>>>>
>>>>>
>>>>>>>>> to the level where you read the source code.
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>> And if you can wait 2+ months for "the book" then that should help as
>>>>>>>>
>>>>>>>>
>>>>> well.
>>>>>
>>>>>
>>>>>>>> :D
>>>>>>>> -MC
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From jmesquita at freeswitch.org  Sun May 30 20:56:08 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 31 May 2010 00:56:08 -0300
Subject: [Freeswitch-users] FSComm linker issue
In-Reply-To: <AANLkTik0SYCbRoOdMfYAW9seeUwcWxiDXqoFZ0ZgwhFY@mail.gmail.com>
References: <AANLkTiliaONCmhMhVsgiXi05v5dIwASq7Zb2t-QJf1YX@mail.gmail.com>
	<AANLkTimnGnnxiWW2DmMTujlOHalpjdHhOOZTMxx_DAB2@mail.gmail.com>
	<AANLkTik0SYCbRoOdMfYAW9seeUwcWxiDXqoFZ0ZgwhFY@mail.gmail.com>
Message-ID: <AANLkTik7rYQuBHFbB4uL6h1X6Z91BN73SM5YK3Kdtdf-@mail.gmail.com>

Thanks to Jeff Lenk, VS proj is now updated.

Thank you once again Jeff.

Jingwei, you might trying it out now?

Regards,
Jo?o Mesquita


On Fri, May 28, 2010 at 12:05 PM, Jingwei Yang <jingwei.yang at gmail.com>wrote:

> Hi Jo?o,
>
> Thanks a lot for the reply. Here's the jira I created:
> http://jira.freeswitch.org/browse/FSCOMM-8.
>
> Frankly speaking, I'm a total newbie to both c++ and VS2008 (just had it
> installed several days back). I might not have the sufficient knowledge to
> contribute. However, if you need any help, please feel free to let me know.
> I'll see what I can do.
>
> Regards,
> -Jingwei
>
> 2010/5/28 Jo?o Mesquita <jmesquita at freeswitch.org>
>
> I must have missed that other email, I am sorry about that. The problem is
>> that the VS Proj was not update with the latest code modifications I've
>> made. I am the sole developer of FSComm and I develop/test in Mac/Linux.
>> It's quite hard for me to keep up with all platforms by myself. Do you feel
>> you could contribute with your MS knowledge? If you feel you can, I can
>> provide you the pointers of what files were added and let you know.
>>
>> Otherwise, could you open a Jira so I won't forget to get to that when I
>> have the time?
>>
>> Thank you for your interest on the project.
>>
>> Regards,
>>  Jo?o Mesquita
>>
>>
>> On Thu, May 27, 2010 at 7:35 AM, Jingwei Yang <jingwei.yang at gmail.com>wrote:
>>
>>> Hello,
>>>
>>> I encountered six linker errors when compiling the latest codes of FSComm
>>> in VS2008. I followed the steps from here:
>>> http://wiki.freeswitch.org/wiki/FSComm#Windows. Someone else seemed to
>>> have the same issue but got no answers:
>>> http://article.gmane.org/gmane.comp.telephony.freeswitch.user/21757/match=fscomm
>>> .
>>>
>>> Please enlighten me how to deal with these errors.
>>>
>>> Thanks and best regards,
>>> -Jingwei
>>>
>>> 10>fshost.obj : error LNK2019: unresolved external symbol "public:
>>> __thiscall Channel::Channel(class QString)" (??0Channel@@QAE at VQString@@@Z)
>>> referenced in function "private: void __thiscall
>>> FSHost::eventChannelCreate(class QSharedPointer<struct switch_event>,class
>>> QString)" (?eventChannelCreate at FSHost
>>> @@AAEXV?$QSharedPointer at Uswitch_event@@@@VQString@@@Z)
>>>
>>> 10>mainwindow.obj : error LNK2019: unresolved external symbol "public:
>>> __thiscall StateDebugDialog::StateDebugDialog(class QWidget *)"
>>> (??0StateDebugDialog@@QAE at PAVQWidget@@@Z) referenced in function
>>> "private: void __thiscall MainWindow::debugEventsTriggered(void)"
>>> (?debugEventsTriggered at MainWindow@@AAEXXZ)
>>>
>>> 10>mainwindow.obj : error LNK2019: unresolved external symbol "public:
>>> __thiscall ConsoleWindow::ConsoleWindow(class QWidget *)" (??0ConsoleWindow@
>>> @QAE at PAVQWidget@@@Z) referenced in function "private: void __thiscall
>>> MainWindow::debugConsoleTriggered(void)" (?debugConsoleTriggered at MainWindow
>>> @@AAEXXZ)
>>>
>>> 10>prefdialog.obj : error LNK2019: unresolved external symbol "public:
>>> __thiscall CodecWidget::CodecWidget(class QWidget *)" (??0CodecWidget@
>>> @QAE at PAVQWidget@@@Z) referenced in function "public: void __thiscall
>>> Ui_PrefDialog::setupUi(class QDialog *)" (?setupUi at Ui_PrefDialog
>>> @@QAEXPAVQDialog@@@Z)
>>>
>>> 10>prefsofia.obj : error LNK2019: unresolved external symbol "public:
>>> void __thiscall CodecWidget::setCodecString(class QString)"
>>> (?setCodecString at CodecWidget@@QAEXVQString@@@Z) referenced in function
>>> "public: void __thiscall PrefSofia::readConfig(void)" (?readConfig at PrefSofia
>>> @@QAEXXZ)
>>>
>>> 10>prefsofia.obj : error LNK2019: unresolved external symbol "public:
>>> class QString __thiscall CodecWidget::getCodecString(void)"
>>> (?getCodecString at CodecWidget@@QAE?AVQString@@XZ) referenced in function
>>> "public: void __thiscall PrefSofia::writeConfig(void)"
>>> (?writeConfig at PrefSofia@@QAEXXZ)
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From saeedahmad1981 at gmail.com  Mon May 31 01:10:27 2010
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Mon, 31 May 2010 10:10:27 +0200
Subject: [Freeswitch-users] FS as Media Gateway Only
In-Reply-To: <DDABF8BB-CB0B-4AE1-9504-8F87CD894D59@gmail.com>
References: <AANLkTimgKRtL06xmyjXUPnpNIjlYkHC6BytzmTqKmVm7@mail.gmail.com>
	<AANLkTikoxsXFuYBxsXXtB26Y9_8f1I1wGmE-u-I_l9PP@mail.gmail.com>
	<AANLkTikgol1TigK9G9vvk90ogtjqeJ3nwPXqdFUXcV5W@mail.gmail.com>
	<BE75B7EE-B77D-4710-899E-AE9E771A28F7@gmail.com>
	<AANLkTimQUri22GlLok8z8LGSFRmufCIXdWGdA4NStbfI@mail.gmail.com>
	<AANLkTimI28Pi_oRNRkgNtT4dT9_cLTPJ9a2XVkgGFzAW@mail.gmail.com>
	<AANLkTimBbGprWu53lQqWhaTmLRdh25l-yCmYbbT1MXBq@mail.gmail.com>
	<AANLkTimSpVOYVB2BiKZWVr3dNrjy0ufQv7YsqAAbdRxS@mail.gmail.com>
	<AANLkTim5o5dc84N4Ag9PenNKvVPfrfr2cQWIQmjVhWuT@mail.gmail.com>
	<DDABF8BB-CB0B-4AE1-9504-8F87CD894D59@gmail.com>
Message-ID: <AANLkTik_o7Zdo_xiNf7u5CYEEZ5EvYHNKXRo1UnzhGcL@mail.gmail.com>

Yup but central billing on RS?

On May 30, 2010 9:05 PM, "David Ponzone" <david.ponzone at gmail.com> wrote:

Well, I think I have a way to solve that.

On RS, set a custom SIP header before bridging the call:
X-Route: Supplier-X

And then in the MS dialplan, you just have to route to the right gateway
according to X-Route.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 ...
Le 30/05/2010 ? 13:21, Saeed Ahmed a ?crit :



> Hi,
>
> Actually i am trying to setup the environment to test it real time,
but theoretically it...


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From david.ponzone at gmail.com  Mon May 31 01:21:09 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Mon, 31 May 2010 10:21:09 +0200
Subject: [Freeswitch-users] FS as Media Gateway Only
In-Reply-To: <AANLkTik_o7Zdo_xiNf7u5CYEEZ5EvYHNKXRo1UnzhGcL@mail.gmail.com>
References: <AANLkTimgKRtL06xmyjXUPnpNIjlYkHC6BytzmTqKmVm7@mail.gmail.com>
	<AANLkTikoxsXFuYBxsXXtB26Y9_8f1I1wGmE-u-I_l9PP@mail.gmail.com>
	<AANLkTikgol1TigK9G9vvk90ogtjqeJ3nwPXqdFUXcV5W@mail.gmail.com>
	<BE75B7EE-B77D-4710-899E-AE9E771A28F7@gmail.com>
	<AANLkTimQUri22GlLok8z8LGSFRmufCIXdWGdA4NStbfI@mail.gmail.com>
	<AANLkTimI28Pi_oRNRkgNtT4dT9_cLTPJ9a2XVkgGFzAW@mail.gmail.com>
	<AANLkTimBbGprWu53lQqWhaTmLRdh25l-yCmYbbT1MXBq@mail.gmail.com>
	<AANLkTimSpVOYVB2BiKZWVr3dNrjy0ufQv7YsqAAbdRxS@mail.gmail.com>
	<AANLkTim5o5dc84N4Ag9PenNKvVPfrfr2cQWIQmjVhWuT@mail.gmail.com>
	<DDABF8BB-CB0B-4AE1-9504-8F87CD894D59@gmail.com>
	<AANLkTik_o7Zdo_xiNf7u5CYEEZ5EvYHNKXRo1UnzhGcL@mail.gmail.com>
Message-ID: <D1FD6941-DEE0-4762-90D3-98D304FC76F4@gmail.com>

I dont think billing is an issue.
You can do it on RS or on MS.
You can actually collect on both to check consistency.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 31/05/2010 ? 10:10, Saeed Ahmed a ?crit :

> Yup but central billing on RS?
>
>
>> On May 30, 2010 9:05 PM, "David Ponzone" <david.ponzone at gmail.com>  
>> wrote:
>>
>> Well, I think I have a way to solve that.
>>
>> On RS, set a custom SIP header before bridging the call:
>> X-Route: Supplier-X
>>
>> And then in the MS dialplan, you just have to route to the right  
>> gateway according to X-Route.
>>
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 ...
>>
>> Le 30/05/2010 ? 13:21, Saeed Ahmed a ?crit :
>>
>>
>> > Hi,
>> >
>> > Actually i am trying to setup the environment to test it real  
>> time, but theoretically it...
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jingwei.yang at gmail.com  Mon May 31 01:27:28 2010
From: jingwei.yang at gmail.com (Jingwei Yang)
Date: Mon, 31 May 2010 16:27:28 +0800
Subject: [Freeswitch-users] FSComm linker issue
In-Reply-To: <AANLkTik7rYQuBHFbB4uL6h1X6Z91BN73SM5YK3Kdtdf-@mail.gmail.com>
References: <AANLkTiliaONCmhMhVsgiXi05v5dIwASq7Zb2t-QJf1YX@mail.gmail.com>
	<AANLkTimnGnnxiWW2DmMTujlOHalpjdHhOOZTMxx_DAB2@mail.gmail.com>
	<AANLkTik0SYCbRoOdMfYAW9seeUwcWxiDXqoFZ0ZgwhFY@mail.gmail.com>
	<AANLkTik7rYQuBHFbB4uL6h1X6Z91BN73SM5YK3Kdtdf-@mail.gmail.com>
Message-ID: <AANLkTikqvXAc_duLQn4dv4K2w0Wo-zF_-aitv7JOMsTy@mail.gmail.com>

Thanks Jo?o, it's built without errors now.

Thanks, Jeff.

Regards,
-Jingwei

2010/5/31 Jo?o Mesquita <jmesquita at freeswitch.org>

> Thanks to Jeff Lenk, VS proj is now updated.
>
> Thank you once again Jeff.
>
> Jingwei, you might trying it out now?
>
> Regards,
> Jo?o Mesquita
>
>
>
> On Fri, May 28, 2010 at 12:05 PM, Jingwei Yang <jingwei.yang at gmail.com>wrote:
>
>> Hi Jo?o,
>>
>> Thanks a lot for the reply. Here's the jira I created:
>> http://jira.freeswitch.org/browse/FSCOMM-8.
>>
>> Frankly speaking, I'm a total newbie to both c++ and VS2008 (just had it
>> installed several days back). I might not have the sufficient knowledge to
>> contribute. However, if you need any help, please feel free to let me know.
>> I'll see what I can do.
>>
>> Regards,
>> -Jingwei
>>
>> 2010/5/28 Jo?o Mesquita <jmesquita at freeswitch.org>
>>
>> I must have missed that other email, I am sorry about that. The problem is
>>> that the VS Proj was not update with the latest code modifications I've
>>> made. I am the sole developer of FSComm and I develop/test in Mac/Linux.
>>> It's quite hard for me to keep up with all platforms by myself. Do you feel
>>> you could contribute with your MS knowledge? If you feel you can, I can
>>> provide you the pointers of what files were added and let you know.
>>>
>>> Otherwise, could you open a Jira so I won't forget to get to that when I
>>> have the time?
>>>
>>> Thank you for your interest on the project.
>>>
>>> Regards,
>>>  Jo?o Mesquita
>>>
>>>
>>> On Thu, May 27, 2010 at 7:35 AM, Jingwei Yang <jingwei.yang at gmail.com>wrote:
>>>
>>>> Hello,
>>>>
>>>> I encountered six linker errors when compiling the latest codes of
>>>> FSComm in VS2008. I followed the steps from here:
>>>> http://wiki.freeswitch.org/wiki/FSComm#Windows. Someone else seemed to
>>>> have the same issue but got no answers:
>>>> http://article.gmane.org/gmane.comp.telephony.freeswitch.user/21757/match=fscomm
>>>> .
>>>>
>>>> Please enlighten me how to deal with these errors.
>>>>
>>>> Thanks and best regards,
>>>> -Jingwei
>>>>
>>>> 10>fshost.obj : error LNK2019: unresolved external symbol "public:
>>>> __thiscall Channel::Channel(class QString)" (??0Channel@@QAE at VQString@@@Z)
>>>> referenced in function "private: void __thiscall
>>>> FSHost::eventChannelCreate(class QSharedPointer<struct switch_event>,class
>>>> QString)" (?eventChannelCreate at FSHost
>>>> @@AAEXV?$QSharedPointer at Uswitch_event@@@@VQString@@@Z)
>>>>
>>>> 10>mainwindow.obj : error LNK2019: unresolved external symbol "public:
>>>> __thiscall StateDebugDialog::StateDebugDialog(class QWidget *)"
>>>> (??0StateDebugDialog@@QAE at PAVQWidget@@@Z) referenced in function
>>>> "private: void __thiscall MainWindow::debugEventsTriggered(void)"
>>>> (?debugEventsTriggered at MainWindow@@AAEXXZ)
>>>>
>>>> 10>mainwindow.obj : error LNK2019: unresolved external symbol "public:
>>>> __thiscall ConsoleWindow::ConsoleWindow(class QWidget *)" (??0ConsoleWindow@
>>>> @QAE at PAVQWidget@@@Z) referenced in function "private: void __thiscall
>>>> MainWindow::debugConsoleTriggered(void)" (?debugConsoleTriggered at MainWindow
>>>> @@AAEXXZ)
>>>>
>>>> 10>prefdialog.obj : error LNK2019: unresolved external symbol "public:
>>>> __thiscall CodecWidget::CodecWidget(class QWidget *)" (??0CodecWidget@
>>>> @QAE at PAVQWidget@@@Z) referenced in function "public: void __thiscall
>>>> Ui_PrefDialog::setupUi(class QDialog *)" (?setupUi at Ui_PrefDialog
>>>> @@QAEXPAVQDialog@@@Z)
>>>>
>>>> 10>prefsofia.obj : error LNK2019: unresolved external symbol "public:
>>>> void __thiscall CodecWidget::setCodecString(class QString)"
>>>> (?setCodecString at CodecWidget@@QAEXVQString@@@Z) referenced in function
>>>> "public: void __thiscall PrefSofia::readConfig(void)" (?readConfig at PrefSofia
>>>> @@QAEXXZ)
>>>>
>>>> 10>prefsofia.obj : error LNK2019: unresolved external symbol "public:
>>>> class QString __thiscall CodecWidget::getCodecString(void)"
>>>> (?getCodecString at CodecWidget@@QAE?AVQString@@XZ) referenced in function
>>>> "public: void __thiscall PrefSofia::writeConfig(void)"
>>>> (?writeConfig at PrefSofia@@QAEXXZ)
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jan.berger at video24.no  Mon May 31 02:33:34 2010
From: jan.berger at video24.no (Jan Berger)
Date: Mon, 31 May 2010 11:33:34 +0200
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
In-Reply-To: <AANLkTikSgh4xN9kb3Lha5dNlacgTA9DqrkHqU3qCWZUT@mail.gmail.com>
References: <490aedfdd7c8ac680bbe0d7f68c786de@mail.gmail.com>
	<4BFBE6C2.5000703@gmail.com>
	<AANLkTilsHn9znajzF0Jj-QDu4je9QW7CeymuBYE0SyKa@mail.gmail.com>
	<4BFBEF7C.9020107@gmail.com>
	<9250A9FF05DB4B879DEF6ECE8E3FFAC1@dell9400>
	<AANLkTilU2OwgH0Qk_Qoc3R7ErPO_Z_VRhePusetkRkeW@mail.gmail.com>
	<AANLkTikXWvEB_C3jqOWLHaVcns9-U5Hok-5MqgE_uzzZ@mail.gmail.com>
	<4BFC4A64.5090208@gmail.com>
	<AANLkTinFtVXa7CPou-ShPYIXHq0AktokhptGSEFKBkge@mail.gmail.com>
	<D311BDCB06324D0DB77898FC6FF4B0EB@dell9400>
	<AANLkTilAMe1lRKK4R6Nmr29FzfI-pP3wRa7Foabmd8Dp@mail.gmail.com>
	<4BFD4562.5030804@gmail.com>
	<AANLkTikGQlZ4N3zfYv18COvbubEi-8LSOZxnAy1KC5sP@mail.gmail.com>
	<4BFEE96A.2010003@gmail.com>
	<AANLkTikSgh4xN9kb3Lha5dNlacgTA9DqrkHqU3qCWZUT@mail.gmail.com>
Message-ID: <D110995190DF45D78114CB61426F9946@dell9400>

Bob,

I suggest you write a wiki-article on FS about this, because this email is
soon forgotten.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bob
Coleman
Sent: 30. mai 2010 23:56
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Questions on Building an application for
FreeSWITCH

I have put up an example for the outbound eventsocket in c#, it is
real basic but gives you one approach to using this method. This has
been coupled together by looking at other samples.

There are two files, one is the dialplan ( 810_sample_ivr.xml ) which
should be placed in the conf/dialplan/default folder
The other file is a zip including a library for talking to the event
socket and a simple test listener.

http://www.devassert.com/apps/freeswitch/810_sample_ivr.xml
http://www.devassert.com/apps/freeswitch/freeswitch.es.net.zip

The password on the zip is cluecon

To try the test, have the dialplan in place and run freeswitch,
connect to an extension with a soft phone, eg 1001, dial 810 and
listen.

Assuming you have freeswitch callie prompts installed
sounds\en\us\callie\conference\8000 all will be good.

Sample is in Visual Studio 2008 and uses a threading library written
by another party.

If you have any questions just ask.

On Fri, May 28, 2010 at 9:51 AM, Phone <phone.bytes at gmail.com> wrote:
> Thanks,
>
> Examples and feedback are most helpful!
>
> Bob Coleman wrote:
>> Hi,
>>
>> Will incorporate some threading into the example I am working on for you.
>>
>> The outbound event socket method is very similar to the dialogic
>> environment, I know how you felt though, but by starting small(like
>> just even answering a call) gets you moving pretty quick. Freeswitch
>> is lots of fun to work with, and the guys on here are very
>> supportive!!
>>
>> Bob
>>
>> On Thu, May 27, 2010 at 3:59 AM, Phone <phone.bytes at gmail.com> wrote:
>>
>>> Thanks to all for the most helpful feedback. ?Sharing your approaches
>>> and experiences are a big help. ?I look forward to the upcoming code
>>> samples.
>>>
>>> I was coming from a windows/dialogic environment where I used a library
>>> that allowed me to work on a little higher level. ?For example, I had a
>>> call to "play a file" that took a parameter of whether or not to allow a
>>> dtmf to interrupt. ?There was also a call to "ReadDtmfs" that took
>>> parameters to specify the number of Dtmf's to read, how long to wait for
>>> them, and what terminating character to use. ?I guess that you could
>>> write some scripts or compiled code with these same types of functions
>>> to simplify some of these routine tasks with reusable code?
>>>
>>> Also, the library handled the threading and scheduling with the OS. ?I
>>> am still unclear on handling the events. ?I guess you have a big loop
>>> reading events and then acting on them using the uuid to determine which
>>> call it is and how to deal with the next step of the call? ?Any feedback
>>> on this part of the project?
>>>
>>> Again, Thanks!
>>>
>>> Bob Coleman wrote:
>>>
>>>> Ah sorry, I started with the esl to get an understanding then wrote my
>>>> own socket library(was actually very easy to do), when I mean docs I
>>>> mean the event socket docs. I still think of it as the esl, my
>>>> mistake.
>>>>
>>>> http://wiki.freeswitch.org/wiki/Event_Socket
>>>>
>>>> I started with a codeplex project, that had been abandoned, and then
>>>> once I understood the structure of the event socket language, was able
>>>> to rewrite it to better handle what we were doing.
>>>>
>>>> I also married it up to an old gotdotnet asterisk fast agi project,
>>>> once again abandoned, to allow for the use of asterisk as well, but in
>>>> the end freeswitch won because we could use just one platform.
>>>>
>>>> I am busy writing a small sample app at the moment to demonstrate a
>>>> problem I am trying to solve. Can release that code once sorted. Will
>>>> be in a week or so. Am intending it as a quick way of testing event
>>>> sockets, and trying various commands etc. before commiting to coding
>>>> something.
>>>>
>>>> Bob
>>>>
>>>> On Wed, May 26, 2010 at 2:26 PM, Jan Berger <jan.berger at video24.no>
wrote:
>>>>
>>>>
>>>>> Hi,
>>>>>
>>>>> Do you have some sample code you could share + what docs did you look
at?
>>>>>
>>>>> I would like to write and test some C# using ESL for my own work.
>>>>>
>>>>> Jan
>>>>>
>>>>> -----Original Message-----
>>>>> From: freeswitch-users-bounces at lists.freeswitch.org
>>>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Bob
>>>>> Coleman
>>>>> Sent: 26. mai 2010 04:07
>>>>> To: freeswitch-users at lists.freeswitch.org
>>>>> Subject: Re: [Freeswitch-users] Questions on Building an application
for
>>>>> FreeSWITCH
>>>>>
>>>>> We used c# as the rest of our systems are windows based. The language
>>>>> doesnt matter too much, as long as you know where you are headed, what
>>>>> performance you require, and what platform you are going to be using.
>>>>>
>>>>> Found the ESL so much easier than the dialogic c library we were
using.
>>>>>
>>>>> The docs for the esl are easy to understand, the thing I couldnt get
>>>>> my head around initially was the dialing out, with the dialogic you
>>>>> are in the middle when you dial, ie already on the channel, but with
>>>>> freeswitch you are kind of the third party when you dial, the channel
>>>>> being created by the dialing and handing it off to be worked on. We
>>>>> make the call via an inbound event socket and hand it off to an
>>>>> outbound event socket application via the dialplan.
>>>>>
>>>>> On Wed, May 26, 2010 at 10:08 AM, Phone <phone.bytes at gmail.com> wrote:
>>>>>
>>>>>
>>>>>> Thanks for the info. ?What language did you use?
>>>>>>
>>>>>> Bob Coleman wrote:
>>>>>>
>>>>>>
>>>>>>> Have just recently completed a project to convert an old windows
>>>>>>> dialogic application(vb6) to FreeSWITCH, would reccommend using the
>>>>>>> ESL, was able to map the old dialogic calls to the ESL calls pretty
>>>>>>> easily. We used a mixture of inbound and outbound sockets, as we
have
>>>>>>> people dialing us, not just dialing out etc.
>>>>>>>
>>>>>>> With the dialogic you open a port and make the call and handle the
>>>>>>> dtmf, with freeswitch you create a socket connection to FreeSWITCH
to
>>>>>>> dial the number and then hand it off to an extension for processing
>>>>>>> the dtmf(that is one approach any way)
>>>>>>>
>>>>>>> Bob
>>>>>>>
>>>>>>> On Wed, May 26, 2010 at 6:45 AM, Michael Collins
<msc at freeswitch.org>
>>>>>>>
>>>>>>>
>>>>> wrote:
>>>>>
>>>>>
>>>>>>>> On Tue, May 25, 2010 at 11:38 AM, Jan Berger
<jan.berger at video24.no>
>>>>>>>>
>>>>>>>>
>>>>> wrote:
>>>>>
>>>>>
>>>>>>>>> Actually - before you get "to smart" - may I suggest that you
start
>>>>>>>>> writing
>>>>>>>>> - or improving - the getting started sections of the doc. Address
the
>>>>>>>>> areas
>>>>>>>>> where you struggle and let others benefit from your work.
>>>>>>>>>
>>>>>>>>> I have been through similar issues myself - FS is one of the
easier
>>>>>>>>> projects
>>>>>>>>> to work with once you get under the hood, but you basically need
to
>>>>>>>>>
>>>>>>>>>
>>>>> evolve
>>>>>
>>>>>
>>>>>>>>> to the level where you read the source code.
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>> And if you can wait 2+ months for "the book" then that should help
as
>>>>>>>>
>>>>>>>>
>>>>> well.
>>>>>
>>>>>
>>>>>>>> :D
>>>>>>>> -MC
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

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



From bobc at devassert.com  Mon May 31 02:52:47 2010
From: bobc at devassert.com (Bob Coleman)
Date: Mon, 31 May 2010 21:52:47 +1200
Subject: [Freeswitch-users] Questions on Building an application for
	FreeSWITCH
In-Reply-To: <D110995190DF45D78114CB61426F9946@dell9400>
References: <490aedfdd7c8ac680bbe0d7f68c786de@mail.gmail.com>
	<4BFBE6C2.5000703@gmail.com>
	<AANLkTilsHn9znajzF0Jj-QDu4je9QW7CeymuBYE0SyKa@mail.gmail.com>
	<4BFBEF7C.9020107@gmail.com>
	<9250A9FF05DB4B879DEF6ECE8E3FFAC1@dell9400>
	<AANLkTilU2OwgH0Qk_Qoc3R7ErPO_Z_VRhePusetkRkeW@mail.gmail.com>
	<AANLkTikXWvEB_C3jqOWLHaVcns9-U5Hok-5MqgE_uzzZ@mail.gmail.com>
	<4BFC4A64.5090208@gmail.com>
	<AANLkTinFtVXa7CPou-ShPYIXHq0AktokhptGSEFKBkge@mail.gmail.com>
	<D311BDCB06324D0DB77898FC6FF4B0EB@dell9400>
	<AANLkTilAMe1lRKK4R6Nmr29FzfI-pP3wRa7Foabmd8Dp@mail.gmail.com>
	<4BFD4562.5030804@gmail.com>
	<AANLkTikGQlZ4N3zfYv18COvbubEi-8LSOZxnAy1KC5sP@mail.gmail.com>
	<4BFEE96A.2010003@gmail.com>
	<AANLkTikSgh4xN9kb3Lha5dNlacgTA9DqrkHqU3qCWZUT@mail.gmail.com>
	<D110995190DF45D78114CB61426F9946@dell9400>
Message-ID: <AANLkTillSAX6L4gum_g9SNjP6O8hGEtXTfcIakZT96_D@mail.gmail.com>

Ok

On Mon, May 31, 2010 at 9:33 PM, Jan Berger <jan.berger at video24.no> wrote:
> Bob,
>
> I suggest you write a wiki-article on FS about this, because this email is
> soon forgotten.
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bob
> Coleman
> Sent: 30. mai 2010 23:56
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Questions on Building an application for
> FreeSWITCH
>
> I have put up an example for the outbound eventsocket in c#, it is
> real basic but gives you one approach to using this method. This has
> been coupled together by looking at other samples.
>
> There are two files, one is the dialplan ( 810_sample_ivr.xml ) which
> should be placed in the conf/dialplan/default folder
> The other file is a zip including a library for talking to the event
> socket and a simple test listener.
>
> http://www.devassert.com/apps/freeswitch/810_sample_ivr.xml
> http://www.devassert.com/apps/freeswitch/freeswitch.es.net.zip
>
> The password on the zip is cluecon
>
> To try the test, have the dialplan in place and run freeswitch,
> connect to an extension with a soft phone, eg 1001, dial 810 and
> listen.
>
> Assuming you have freeswitch callie prompts installed
> sounds\en\us\callie\conference\8000 all will be good.
>
> Sample is in Visual Studio 2008 and uses a threading library written
> by another party.
>
> If you have any questions just ask.
>
> On Fri, May 28, 2010 at 9:51 AM, Phone <phone.bytes at gmail.com> wrote:
>> Thanks,
>>
>> Examples and feedback are most helpful!
>>
>> Bob Coleman wrote:
>>> Hi,
>>>
>>> Will incorporate some threading into the example I am working on for you.
>>>
>>> The outbound event socket method is very similar to the dialogic
>>> environment, I know how you felt though, but by starting small(like
>>> just even answering a call) gets you moving pretty quick. Freeswitch
>>> is lots of fun to work with, and the guys on here are very
>>> supportive!!
>>>
>>> Bob
>>>
>>> On Thu, May 27, 2010 at 3:59 AM, Phone <phone.bytes at gmail.com> wrote:
>>>
>>>> Thanks to all for the most helpful feedback. ?Sharing your approaches
>>>> and experiences are a big help. ?I look forward to the upcoming code
>>>> samples.
>>>>
>>>> I was coming from a windows/dialogic environment where I used a library
>>>> that allowed me to work on a little higher level. ?For example, I had a
>>>> call to "play a file" that took a parameter of whether or not to allow a
>>>> dtmf to interrupt. ?There was also a call to "ReadDtmfs" that took
>>>> parameters to specify the number of Dtmf's to read, how long to wait for
>>>> them, and what terminating character to use. ?I guess that you could
>>>> write some scripts or compiled code with these same types of functions
>>>> to simplify some of these routine tasks with reusable code?
>>>>
>>>> Also, the library handled the threading and scheduling with the OS. ?I
>>>> am still unclear on handling the events. ?I guess you have a big loop
>>>> reading events and then acting on them using the uuid to determine which
>>>> call it is and how to deal with the next step of the call? ?Any feedback
>>>> on this part of the project?
>>>>
>>>> Again, Thanks!
>>>>
>>>> Bob Coleman wrote:
>>>>
>>>>> Ah sorry, I started with the esl to get an understanding then wrote my
>>>>> own socket library(was actually very easy to do), when I mean docs I
>>>>> mean the event socket docs. I still think of it as the esl, my
>>>>> mistake.
>>>>>
>>>>> http://wiki.freeswitch.org/wiki/Event_Socket
>>>>>
>>>>> I started with a codeplex project, that had been abandoned, and then
>>>>> once I understood the structure of the event socket language, was able
>>>>> to rewrite it to better handle what we were doing.
>>>>>
>>>>> I also married it up to an old gotdotnet asterisk fast agi project,
>>>>> once again abandoned, to allow for the use of asterisk as well, but in
>>>>> the end freeswitch won because we could use just one platform.
>>>>>
>>>>> I am busy writing a small sample app at the moment to demonstrate a
>>>>> problem I am trying to solve. Can release that code once sorted. Will
>>>>> be in a week or so. Am intending it as a quick way of testing event
>>>>> sockets, and trying various commands etc. before commiting to coding
>>>>> something.
>>>>>
>>>>> Bob
>>>>>
>>>>> On Wed, May 26, 2010 at 2:26 PM, Jan Berger <jan.berger at video24.no>
> wrote:
>>>>>
>>>>>
>>>>>> Hi,
>>>>>>
>>>>>> Do you have some sample code you could share + what docs did you look
> at?
>>>>>>
>>>>>> I would like to write and test some C# using ESL for my own work.
>>>>>>
>>>>>> Jan
>>>>>>
>>>>>> -----Original Message-----
>>>>>> From: freeswitch-users-bounces at lists.freeswitch.org
>>>>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Bob
>>>>>> Coleman
>>>>>> Sent: 26. mai 2010 04:07
>>>>>> To: freeswitch-users at lists.freeswitch.org
>>>>>> Subject: Re: [Freeswitch-users] Questions on Building an application
> for
>>>>>> FreeSWITCH
>>>>>>
>>>>>> We used c# as the rest of our systems are windows based. The language
>>>>>> doesnt matter too much, as long as you know where you are headed, what
>>>>>> performance you require, and what platform you are going to be using.
>>>>>>
>>>>>> Found the ESL so much easier than the dialogic c library we were
> using.
>>>>>>
>>>>>> The docs for the esl are easy to understand, the thing I couldnt get
>>>>>> my head around initially was the dialing out, with the dialogic you
>>>>>> are in the middle when you dial, ie already on the channel, but with
>>>>>> freeswitch you are kind of the third party when you dial, the channel
>>>>>> being created by the dialing and handing it off to be worked on. We
>>>>>> make the call via an inbound event socket and hand it off to an
>>>>>> outbound event socket application via the dialplan.
>>>>>>
>>>>>> On Wed, May 26, 2010 at 10:08 AM, Phone <phone.bytes at gmail.com> wrote:
>>>>>>
>>>>>>
>>>>>>> Thanks for the info. ?What language did you use?
>>>>>>>
>>>>>>> Bob Coleman wrote:
>>>>>>>
>>>>>>>
>>>>>>>> Have just recently completed a project to convert an old windows
>>>>>>>> dialogic application(vb6) to FreeSWITCH, would reccommend using the
>>>>>>>> ESL, was able to map the old dialogic calls to the ESL calls pretty
>>>>>>>> easily. We used a mixture of inbound and outbound sockets, as we
> have
>>>>>>>> people dialing us, not just dialing out etc.
>>>>>>>>
>>>>>>>> With the dialogic you open a port and make the call and handle the
>>>>>>>> dtmf, with freeswitch you create a socket connection to FreeSWITCH
> to
>>>>>>>> dial the number and then hand it off to an extension for processing
>>>>>>>> the dtmf(that is one approach any way)
>>>>>>>>
>>>>>>>> Bob
>>>>>>>>
>>>>>>>> On Wed, May 26, 2010 at 6:45 AM, Michael Collins
> <msc at freeswitch.org>
>>>>>>>>
>>>>>>>>
>>>>>> wrote:
>>>>>>
>>>>>>
>>>>>>>>> On Tue, May 25, 2010 at 11:38 AM, Jan Berger
> <jan.berger at video24.no>
>>>>>>>>>
>>>>>>>>>
>>>>>> wrote:
>>>>>>
>>>>>>
>>>>>>>>>> Actually - before you get "to smart" - may I suggest that you
> start
>>>>>>>>>> writing
>>>>>>>>>> - or improving - the getting started sections of the doc. Address
> the
>>>>>>>>>> areas
>>>>>>>>>> where you struggle and let others benefit from your work.
>>>>>>>>>>
>>>>>>>>>> I have been through similar issues myself - FS is one of the
> easier
>>>>>>>>>> projects
>>>>>>>>>> to work with once you get under the hood, but you basically need
> to
>>>>>>>>>>
>>>>>>>>>>
>>>>>> evolve
>>>>>>
>>>>>>
>>>>>>>>>> to the level where you read the source code.
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>> And if you can wait 2+ months for "the book" then that should help
> as
>>>>>>>>>
>>>>>>>>>
>>>>>> well.
>>>>>>
>>>>>>
>>>>>>>>> :D
>>>>>>>>> -MC
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _______________________________________________
>>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> FreeSWITCH-users mailing list
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From joakim at gissberg.nu  Fri May 28 12:03:19 2010
From: joakim at gissberg.nu (Joakim Gissberg)
Date: Fri, 28 May 2010 22:03:19 +0300
Subject: [Freeswitch-users] Roadwarrier setup on external profile.
Message-ID: <4C001377.9060109@gissberg.nu>

Hi.

I have set up my freeswitch server so that the default profile binds to 
my internal interface port 5060, and the external profile is bind to my 
external interface on the same port.
What I want to accomplis is that each incomming call also rings the 
"roadwarrior" connected sip-clients, ie the ones authenticated to the 
external profile.

I have created a ringgroup, and each incoming call is ringed in the 
public context (dialplan/public/00_XXX_incoming.xml) with

> <action application="transfer" data="1020 XML default"/>

and every user is in that group. Works if I authenticate with the 
internal profile.

I have uncomment the 
transfer-authentificated-sip-clients-to-default-context part in 
dialplan/public.xml and I can call and use those extensions from the 
external profile (for example call a local extensions authenticated 
against the default internal profile from the roadwarrior).
The problem firstly I guess is that an internal client can't call the 
externaly connected client, the call gets directly to voicemail with 
error "[ERR] switch_ivr_originate.c:2493 Cannot create outgoing channel 
of type [error] cause: [USER_NOT_REGISTERED]"

The configuration is pretty much default other than the changes 
explained above and added outgoing SIP-trunks.

Any pointers to how I could solve this?

  - J



From ewin.hogan at gmail.com  Fri May 28 15:44:46 2010
From: ewin.hogan at gmail.com (blacked)
Date: Fri, 28 May 2010 15:44:46 -0700 (PDT)
Subject: [Freeswitch-users] RE FER with Replaces
Message-ID: <28712663.post@talk.nabble.com>


hi everyone well about my last post now I'll be more specific did anyone know
how can I make a transfer, a REFER with replaces to repleace FS in the loop
of the call and let the two legs with out FS.

hope someone can help me
-- 
View this message in context: http://old.nabble.com/REFER-with-Replaces-tp28712663p28712663.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From peter.olsson at visionutveckling.se  Mon May 31 06:10:12 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 31 May 2010 15:10:12 +0200
Subject: [Freeswitch-users] RE FER with Replaces
In-Reply-To: <28712663.post@talk.nabble.com>
References: <28712663.post@talk.nabble.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C567C9C8B2C@cooper>

Use uuid_simplify on the bridged call, it will do exactly this.

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r blacked
Skickat: den 29 maj 2010 00:45
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] RE FER with Replaces


hi everyone well about my last post now I'll be more specific did anyone know
how can I make a transfer, a REFER with replaces to repleace FS in the loop
of the call and let the two legs with out FS.

hope someone can help me
-- 
View this message in context: http://old.nabble.com/REFER-with-Replaces-tp28712663p28712663.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.


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

!DSPAM:4c03b43232931738480613!



From mcampbellsmith at gmail.com  Mon May 31 07:12:48 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Mon, 31 May 2010 16:12:48 +0200
Subject: [Freeswitch-users] rtpmap question PCMU/PCMA
Message-ID: <AANLkTiloOAaoCWFD74aZQYuRex-kRCTG0zWLkV927jqu@mail.gmail.com>

Hi All,

I'm sure I've discussed this before, but I searched through my gmail
and google and couldn't find the answer.

Below is the SDP parameters from my sip provider.  Is it mandatory to
always include the rtpmap details for PCMU/PCMA codes?  For example
something like 'a=rtpmap:8 PCMA/8000' and 'a=rtpmap:0 PCMU/8000'?

I'm using sipdroid on android and it rejects this with 'codec not supported'

Thanks



o=Sippy 141730476 0 IN IP4 xxx.xxx.xxx.xxx
s=-
t=0 0
m=audio 47676 RTP/AVP 8 0 18 101
c=IN IP4 xxx.xxx.xxx.xxx
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sqn: 0
a=cdsc: 1 audio RTP/AVP  8
a=cdsc: 2 image udptl t38
a=cpar: a=T38FaxUdpEC:t38UDPRedundancy
a=cpar: a=T38FaxVersion:0
a=cpar: a=T38MaxBitRate:14400
]


From david.ponzone at gmail.com  Mon May 31 07:29:43 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Mon, 31 May 2010 16:29:43 +0200
Subject: [Freeswitch-users] rtpmap question PCMU/PCMA
In-Reply-To: <AANLkTiloOAaoCWFD74aZQYuRex-kRCTG0zWLkV927jqu@mail.gmail.com>
References: <AANLkTiloOAaoCWFD74aZQYuRex-kRCTG0zWLkV927jqu@mail.gmail.com>
Message-ID: <0495C809-A9F9-425D-ACF1-1B706324A1AF@gmail.com>

Mark,

This looks like a T38 Re-INVITE, but a weird one.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au  
titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si  
vous n'?tes pas destinataire de ce message, merci de le d?truire  
imm?diatement et d'avertir l'exp?diteur.




Le 31/05/2010 ? 16:12, Mark Campbell-Smith a ?crit :

> Hi All,
>
> I'm sure I've discussed this before, but I searched through my gmail
> and google and couldn't find the answer.
>
> Below is the SDP parameters from my sip provider.  Is it mandatory to
> always include the rtpmap details for PCMU/PCMA codes?  For example
> something like 'a=rtpmap:8 PCMA/8000' and 'a=rtpmap:0 PCMU/8000'?
>
> I'm using sipdroid on android and it rejects this with 'codec not  
> supported'
>
> Thanks
>
>
>
> o=Sippy 141730476 0 IN IP4 xxx.xxx.xxx.xxx
> s=-
> t=0 0
> m=audio 47676 RTP/AVP 8 0 18 101
> c=IN IP4 xxx.xxx.xxx.xxx
> a=fmtp:18 annexb=yes
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=sqn: 0
> a=cdsc: 1 audio RTP/AVP  8
> a=cdsc: 2 image udptl t38
> a=cpar: a=T38FaxUdpEC:t38UDPRedundancy
> a=cpar: a=T38FaxVersion:0
> a=cpar: a=T38MaxBitRate:14400
> ]
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
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From mcampbellsmith at gmail.com  Mon May 31 07:57:16 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Mon, 31 May 2010 16:57:16 +0200
Subject: [Freeswitch-users] rtpmap question PCMU/PCMA
In-Reply-To: <0495C809-A9F9-425D-ACF1-1B706324A1AF@gmail.com>
References: <AANLkTiloOAaoCWFD74aZQYuRex-kRCTG0zWLkV927jqu@mail.gmail.com>
	<0495C809-A9F9-425D-ACF1-1B706324A1AF@gmail.com>
Message-ID: <AANLkTil6MLm29u_tB-BXL_V-geDrn0_1CUmlmDHGZ1N2@mail.gmail.com>

Hi David,

Its an INVITE.  Full invite below:

   INVITE sip:gw+Phonzo at 124.xxx.xxx.xx:5080;transport=udp;gw=Phonzo SIP/2.0
   Record-Route: <sip:80.232.37.178;ftag=262787ae2a9104a0c7700794a69028aco;lr>
   Via: SIP/2.0/UDP
80.232.37.178;branch=z9hG4bK8a48.ecc54191b911cf6bfd73daeefeae0ada.0
   Via: SIP/2.0/UDP
80.232.37.178:5061;branch=z9hG4bK443b032c2a924807acf39718946e2c9e;rport=5061
   Max-Forwards: 16
   From: 010711xxxx
<sip:711xxxx at 80.232.37.178>;tag=262787ae2a9104a0c7700794a69028aco
   To: <sip:4610xxxxxxx at 80.232.37.178>
   Call-ID: M2Q1NjhhNmZjOGJjMDc3ODhlNzUyYzRiM2ZkMjQyZTE.
   CSeq: 200 INVITE
   Contact: Anonymous <sip:80.232.37.178:5061>
   Expires: 300
   User-Agent: Sippy
   cisco-GUID: 1658214937-1822691807-2631794736-96895450
   h323-conf-id: 1658214937-1822691807-2631794736-96895450
   Content-disposition: session
   Content-Length: 364
   Content-Type: application/sdp

   v=0
   o=Sippy 141730476 0 IN IP4 80.232.37.178
   s=-
   t=0 0
   m=audio 47676 RTP/AVP 8 0 18 101
   c=IN IP4 213.50.91.3
   a=fmtp:18 annexb=yes
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   a=sqn: 0
   a=cdsc: 1 audio RTP/AVP  8
   a=cdsc: 2 image udptl t38
   a=cpar: a=T38FaxUdpEC:t38UDPRedundancy
   a=cpar: a=T38FaxVersion:0
   a=cpar: a=T38MaxBitRate:14400
   a=sendrecv

Regards
Mark

On Mon, May 31, 2010 at 4:29 PM, David Ponzone <david.ponzone at gmail.com> wrote:
> Mark,
> This looks like a T38 Re-INVITE, but a weird one.
> David Ponzone ?Direction Technique
> email: david.ponzone at ipeva.fr
> tel: ? ? ?01 74 03 18 97
> gsm: ? 06 66 98 76 34
> Service Client?IPeva
> tel: ? ? ?0811 46 26 26
> www.ipeva.fr? -? ?www.ipeva-studio.com
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
> Le 31/05/2010 ? 16:12, Mark Campbell-Smith a ?crit :
>
> Hi All,
>
> I'm sure I've discussed this before, but I searched through my gmail
> and google and couldn't find the answer.
>
> Below is the SDP parameters from my sip provider. ?Is it mandatory to
> always include the rtpmap details for PCMU/PCMA codes? ?For example
> something like 'a=rtpmap:8 PCMA/8000' and 'a=rtpmap:0 PCMU/8000'?
>
> I'm using sipdroid on android and it rejects this with 'codec not supported'
>
> Thanks
>
>
>
> o=Sippy 141730476 0 IN IP4 xxx.xxx.xxx.xxx
> s=-
> t=0 0
> m=audio 47676 RTP/AVP 8 0 18 101
> c=IN IP4 xxx.xxx.xxx.xxx
> a=fmtp:18 annexb=yes
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=sqn: 0
> a=cdsc: 1 audio RTP/AVP ?8
> a=cdsc: 2 image udptl t38
> a=cpar: a=T38FaxUdpEC:t38UDPRedundancy
> a=cpar: a=T38FaxVersion:0
> a=cpar: a=T38MaxBitRate:14400
> ]
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From david.ponzone at gmail.com  Mon May 31 08:12:11 2010
From: david.ponzone at gmail.com (David Ponzone)
Date: Mon, 31 May 2010 17:12:11 +0200
Subject: [Freeswitch-users] rtpmap question PCMU/PCMA
In-Reply-To: <AANLkTil6MLm29u_tB-BXL_V-geDrn0_1CUmlmDHGZ1N2@mail.gmail.com>
References: <AANLkTiloOAaoCWFD74aZQYuRex-kRCTG0zWLkV927jqu@mail.gmail.com>
	<0495C809-A9F9-425D-ACF1-1B706324A1AF@gmail.com>
	<AANLkTil6MLm29u_tB-BXL_V-geDrn0_1CUmlmDHGZ1N2@mail.gmail.com>
Message-ID: <E56A9939-B461-4B2F-8E3D-0E4FDF194055@gmail.com>

Ah ok.
Are you sure your device is not choking because of the T38-related  
lines in the SDP ?
Until now, all the devices I used only advertise T38 in the Re-INVITE,  
and I think (to be confirmed) that's the "standard" way to do it.
I guess there could be a reason not to advertise it in the initial  
INVITE.

As for the missing rtpmap lines, I don't know if it's valid not to  
send them.
I guess it also relies on the device at the other end.

Can you confirm how is this related to FS ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis  
? l'intention exclusive de ses destinataires. Toute utilisation ou  
diffusion non autoris?e est interdite. Tout message ?lectronique est  
susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre  
de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes  
pas destinataire de ce message, merci de le d?truire imm?diatement et  
d'avertir l'exp?diteur.




Le 31/05/2010 ? 16:57, Mark Campbell-Smith a ?crit :

> Hi David,
>
> Its an INVITE.  Full invite below:
>
>   INVITE sip:gw+Phonzo at 124.xxx.xxx.xx:5080;transport=udp;gw=Phonzo  
> SIP/2.0
>   Record-Route: <sip:80.232.37.178;ftag=262787ae2a9104a0c7700794a69028aco;lr 
> >
>   Via: SIP/2.0/UDP
> 80.232.37.178;branch=z9hG4bK8a48.ecc54191b911cf6bfd73daeefeae0ada.0
>   Via: SIP/2.0/UDP
> 80.232.37.178 
> :5061;branch=z9hG4bK443b032c2a924807acf39718946e2c9e;rport=5061
>   Max-Forwards: 16
>   From: 010711xxxx
> <sip:711xxxx at 80.232.37.178>;tag=262787ae2a9104a0c7700794a69028aco
>   To: <sip:4610xxxxxxx at 80.232.37.178>
>   Call-ID: M2Q1NjhhNmZjOGJjMDc3ODhlNzUyYzRiM2ZkMjQyZTE.
>   CSeq: 200 INVITE
>   Contact: Anonymous <sip:80.232.37.178:5061>
>   Expires: 300
>   User-Agent: Sippy
>   cisco-GUID: 1658214937-1822691807-2631794736-96895450
>   h323-conf-id: 1658214937-1822691807-2631794736-96895450
>   Content-disposition: session
>   Content-Length: 364
>   Content-Type: application/sdp
>
>   v=0
>   o=Sippy 141730476 0 IN IP4 80.232.37.178
>   s=-
>   t=0 0
>   m=audio 47676 RTP/AVP 8 0 18 101
>   c=IN IP4 213.50.91.3
>   a=fmtp:18 annexb=yes
>   a=rtpmap:101 telephone-event/8000
>   a=fmtp:101 0-15
>   a=sqn: 0
>   a=cdsc: 1 audio RTP/AVP  8
>   a=cdsc: 2 image udptl t38
>   a=cpar: a=T38FaxUdpEC:t38UDPRedundancy
>   a=cpar: a=T38FaxVersion:0
>   a=cpar: a=T38MaxBitRate:14400
>   a=sendrecv
>
> Regards
> Mark
>
> On Mon, May 31, 2010 at 4:29 PM, David Ponzone <david.ponzone at gmail.com 
> > wrote:
>> Mark,
>> This looks like a T38 Re-INVITE, but a weird one.
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> Ce message et toutes les pi?ces jointes sont confidentiels et  
>> ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou  
>> diffusion
>> non autoris?e est interdite. Tout message ?lectronique est  
>> susceptible
>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce  
>> message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire  
>> de ce
>> message, merci de le d?truire imm?diatement et d'avertir  
>> l'exp?diteur.
>>
>>
>>
>> Le 31/05/2010 ? 16:12, Mark Campbell-Smith a ?crit :
>>
>> Hi All,
>>
>> I'm sure I've discussed this before, but I searched through my gmail
>> and google and couldn't find the answer.
>>
>> Below is the SDP parameters from my sip provider.  Is it mandatory to
>> always include the rtpmap details for PCMU/PCMA codes?  For example
>> something like 'a=rtpmap:8 PCMA/8000' and 'a=rtpmap:0 PCMU/8000'?
>>
>> I'm using sipdroid on android and it rejects this with 'codec not  
>> supported'
>>
>> Thanks
>>
>>
>>
>> o=Sippy 141730476 0 IN IP4 xxx.xxx.xxx.xxx
>> s=-
>> t=0 0
>> m=audio 47676 RTP/AVP 8 0 18 101
>> c=IN IP4 xxx.xxx.xxx.xxx
>> a=fmtp:18 annexb=yes
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>> a=sqn: 0
>> a=cdsc: 1 audio RTP/AVP  8
>> a=cdsc: 2 image udptl t38
>> a=cpar: a=T38FaxUdpEC:t38UDPRedundancy
>> a=cpar: a=T38FaxVersion:0
>> a=cpar: a=T38MaxBitRate:14400
>> ]
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From ewin.hogan at gmail.com  Mon May 31 08:54:24 2010
From: ewin.hogan at gmail.com (Black Star)
Date: Mon, 31 May 2010 10:54:24 -0500
Subject: [Freeswitch-users] RE FER with Replaces
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C567C9C8B2C@cooper>
References: <28712663.post@talk.nabble.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C567C9C8B2C@cooper>
Message-ID: <AANLkTil01_notTAD9lk7VBNJ_vMClYYgNh2uKL0V4rWV@mail.gmail.com>

sorry if I missunderstood some concepts on a call but I originate two legs
one of that legs isn't registered in FS, so I have two channels, then I
bridge the legs and then I have the same two channels and a call, so then I
use uuid_simplify and the call it's gone but I still have the two
channels... it's that right? I looking for a kind of deflect that I have one
leg and I deflect to for exaple to "sofia/internal/example at 192.168.1.XXX"
and then the call it's gone and also the channels I used too. well hope some
one can help me :)

2010/5/31 Peter Olsson <peter.olsson at visionutveckling.se>

> Use uuid_simplify on the bridged call, it will do exactly this.
>
> /Peter
>
>
> -----Ursprungligt meddelande-----
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] F?r blacked
> Skickat: den 29 maj 2010 00:45
> Till: freeswitch-users at lists.freeswitch.org
> ?mne: [Freeswitch-users] RE FER with Replaces
>
>
> hi everyone well about my last post now I'll be more specific did anyone
> know
> how can I make a transfer, a REFER with replaces to repleace FS in the loop
> of the call and let the two legs with out FS.
>
> hope someone can help me
> --
> View this message in context:
> http://old.nabble.com/REFER-with-Replaces-tp28712663p28712663.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> !DSPAM:4c03b43232931738480613!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jmesquita at freeswitch.org  Mon May 31 09:12:08 2010
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 31 May 2010 13:12:08 -0300
Subject: [Freeswitch-users] FSComm linker issue
In-Reply-To: <AANLkTikqvXAc_duLQn4dv4K2w0Wo-zF_-aitv7JOMsTy@mail.gmail.com>
References: <AANLkTiliaONCmhMhVsgiXi05v5dIwASq7Zb2t-QJf1YX@mail.gmail.com>
	<AANLkTimnGnnxiWW2DmMTujlOHalpjdHhOOZTMxx_DAB2@mail.gmail.com>
	<AANLkTik0SYCbRoOdMfYAW9seeUwcWxiDXqoFZ0ZgwhFY@mail.gmail.com>
	<AANLkTik7rYQuBHFbB4uL6h1X6Z91BN73SM5YK3Kdtdf-@mail.gmail.com>
	<AANLkTikqvXAc_duLQn4dv4K2w0Wo-zF_-aitv7JOMsTy@mail.gmail.com>
Message-ID: <AANLkTikdUdloVFJW9Oa1tiiD6mSz1TGaDpTgeetxaMqq@mail.gmail.com>

No, please, we should thank Jeff, not me.

Please, report anything you find on Jira and GUI improvements/suggestions
are welcome.

Regards,
Jo?o Mesquita


On Mon, May 31, 2010 at 5:27 AM, Jingwei Yang <jingwei.yang at gmail.com>wrote:

> Thanks Jo?o, it's built without errors now.
>
> Thanks, Jeff.
>
> Regards,
> -Jingwei
>
> 2010/5/31 Jo?o Mesquita <jmesquita at freeswitch.org>
>
> Thanks to Jeff Lenk, VS proj is now updated.
>>
>> Thank you once again Jeff.
>>
>> Jingwei, you might trying it out now?
>>
>> Regards,
>> Jo?o Mesquita
>>
>>
>>
>> On Fri, May 28, 2010 at 12:05 PM, Jingwei Yang <jingwei.yang at gmail.com>wrote:
>>
>>> Hi Jo?o,
>>>
>>> Thanks a lot for the reply. Here's the jira I created:
>>> http://jira.freeswitch.org/browse/FSCOMM-8.
>>>
>>> Frankly speaking, I'm a total newbie to both c++ and VS2008 (just had it
>>> installed several days back). I might not have the sufficient knowledge to
>>> contribute. However, if you need any help, please feel free to let me know.
>>> I'll see what I can do.
>>>
>>> Regards,
>>> -Jingwei
>>>
>>> 2010/5/28 Jo?o Mesquita <jmesquita at freeswitch.org>
>>>
>>> I must have missed that other email, I am sorry about that. The problem
>>>> is that the VS Proj was not update with the latest code modifications I've
>>>> made. I am the sole developer of FSComm and I develop/test in Mac/Linux.
>>>> It's quite hard for me to keep up with all platforms by myself. Do you feel
>>>> you could contribute with your MS knowledge? If you feel you can, I can
>>>> provide you the pointers of what files were added and let you know.
>>>>
>>>> Otherwise, could you open a Jira so I won't forget to get to that when I
>>>> have the time?
>>>>
>>>> Thank you for your interest on the project.
>>>>
>>>> Regards,
>>>>  Jo?o Mesquita
>>>>
>>>>
>>>> On Thu, May 27, 2010 at 7:35 AM, Jingwei Yang <jingwei.yang at gmail.com>wrote:
>>>>
>>>>> Hello,
>>>>>
>>>>> I encountered six linker errors when compiling the latest codes of
>>>>> FSComm in VS2008. I followed the steps from here:
>>>>> http://wiki.freeswitch.org/wiki/FSComm#Windows. Someone else seemed to
>>>>> have the same issue but got no answers:
>>>>> http://article.gmane.org/gmane.comp.telephony.freeswitch.user/21757/match=fscomm
>>>>> .
>>>>>
>>>>> Please enlighten me how to deal with these errors.
>>>>>
>>>>> Thanks and best regards,
>>>>> -Jingwei
>>>>>
>>>>> 10>fshost.obj : error LNK2019: unresolved external symbol "public:
>>>>> __thiscall Channel::Channel(class QString)" (??0Channel@@QAE at VQString@@@Z)
>>>>> referenced in function "private: void __thiscall
>>>>> FSHost::eventChannelCreate(class QSharedPointer<struct switch_event>,class
>>>>> QString)" (?eventChannelCreate at FSHost
>>>>> @@AAEXV?$QSharedPointer at Uswitch_event@@@@VQString@@@Z)
>>>>>
>>>>> 10>mainwindow.obj : error LNK2019: unresolved external symbol "public:
>>>>> __thiscall StateDebugDialog::StateDebugDialog(class QWidget *)"
>>>>> (??0StateDebugDialog@@QAE at PAVQWidget@@@Z) referenced in function
>>>>> "private: void __thiscall MainWindow::debugEventsTriggered(void)"
>>>>> (?debugEventsTriggered at MainWindow@@AAEXXZ)
>>>>>
>>>>> 10>mainwindow.obj : error LNK2019: unresolved external symbol "public:
>>>>> __thiscall ConsoleWindow::ConsoleWindow(class QWidget *)" (??0ConsoleWindow@
>>>>> @QAE at PAVQWidget@@@Z) referenced in function "private: void __thiscall
>>>>> MainWindow::debugConsoleTriggered(void)" (?debugConsoleTriggered at MainWindow
>>>>> @@AAEXXZ)
>>>>>
>>>>> 10>prefdialog.obj : error LNK2019: unresolved external symbol "public:
>>>>> __thiscall CodecWidget::CodecWidget(class QWidget *)" (??0CodecWidget@
>>>>> @QAE at PAVQWidget@@@Z) referenced in function "public: void __thiscall
>>>>> Ui_PrefDialog::setupUi(class QDialog *)" (?setupUi at Ui_PrefDialog
>>>>> @@QAEXPAVQDialog@@@Z)
>>>>>
>>>>> 10>prefsofia.obj : error LNK2019: unresolved external symbol "public:
>>>>> void __thiscall CodecWidget::setCodecString(class QString)"
>>>>> (?setCodecString at CodecWidget@@QAEXVQString@@@Z) referenced in function
>>>>> "public: void __thiscall PrefSofia::readConfig(void)" (?readConfig at PrefSofia
>>>>> @@QAEXXZ)
>>>>>
>>>>> 10>prefsofia.obj : error LNK2019: unresolved external symbol "public:
>>>>> class QString __thiscall CodecWidget::getCodecString(void)"
>>>>> (?getCodecString at CodecWidget@@QAE?AVQString@@XZ) referenced in
>>>>> function "public: void __thiscall PrefSofia::writeConfig(void)"
>>>>> (?writeConfig at PrefSofia@@QAEXXZ)
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lands at freenet.de  Mon May 31 09:29:19 2010
From: lands at freenet.de (lands at freenet.de)
Date: Mon, 31 May 2010 18:29:19 +0200
Subject: [Freeswitch-users] anti SPIT mechanisms
Message-ID: <4C03E3DF.3070603@freenet.de>

Hi,

today I got my first automated SPIT call over my ekiga account. So I
looked for anti SPIT mechanisms in freeswitch?s wiki and found
http://wiki.freeswitch.org/wiki/Mod_limit

Well, it only limits spit calls. So I looked further and found that
asterisk script:


exten => 123456,1,Set(pin=${UNIQUEID:-2})
exten => 123456,n,Playback(your_password_is)
exten => 123456,n,SayDigits(${pin})
exten => 123456,n,Authenticate(${pin})
exten => 123456,n,Goto(some_context_ring_ring,s,1)


So, let the caller repeat a random number and has to prove that he/she
is an human. Well, has anyone done something like this or better
mechanisms within freeswitch?

Chris


From mike at jerris.com  Mon May 31 09:35:54 2010
From: mike at jerris.com (Michael Jerris)
Date: Mon, 31 May 2010 12:35:54 -0400
Subject: [Freeswitch-users] rtpmap question PCMU/PCMA
In-Reply-To: <AANLkTil6MLm29u_tB-BXL_V-geDrn0_1CUmlmDHGZ1N2@mail.gmail.com>
References: <AANLkTiloOAaoCWFD74aZQYuRex-kRCTG0zWLkV927jqu@mail.gmail.com>
	<0495C809-A9F9-425D-ACF1-1B706324A1AF@gmail.com>
	<AANLkTil6MLm29u_tB-BXL_V-geDrn0_1CUmlmDHGZ1N2@mail.gmail.com>
Message-ID: <088D4AC5-BE1D-4304-8139-FB00CFC796F9@jerris.com>

This seems to be a badly broken sdp attempting to offer audio and t.38 but missing the m=image line from the sdp.

Mike

On May 31, 2010, at 10:57 AM, Mark Campbell-Smith wrote:

> Hi David,
> 
> Its an INVITE.  Full invite below:
> 
>   INVITE sip:gw+Phonzo at 124.xxx.xxx.xx:5080;transport=udp;gw=Phonzo SIP/2.0
>   Record-Route: <sip:80.232.37.178;ftag=262787ae2a9104a0c7700794a69028aco;lr>
>   Via: SIP/2.0/UDP
> 80.232.37.178;branch=z9hG4bK8a48.ecc54191b911cf6bfd73daeefeae0ada.0
>   Via: SIP/2.0/UDP
> 80.232.37.178:5061;branch=z9hG4bK443b032c2a924807acf39718946e2c9e;rport=5061
>   Max-Forwards: 16
>   From: 010711xxxx
> <sip:711xxxx at 80.232.37.178>;tag=262787ae2a9104a0c7700794a69028aco
>   To: <sip:4610xxxxxxx at 80.232.37.178>
>   Call-ID: M2Q1NjhhNmZjOGJjMDc3ODhlNzUyYzRiM2ZkMjQyZTE.
>   CSeq: 200 INVITE
>   Contact: Anonymous <sip:80.232.37.178:5061>
>   Expires: 300
>   User-Agent: Sippy
>   cisco-GUID: 1658214937-1822691807-2631794736-96895450
>   h323-conf-id: 1658214937-1822691807-2631794736-96895450
>   Content-disposition: session
>   Content-Length: 364
>   Content-Type: application/sdp
> 
>   v=0
>   o=Sippy 141730476 0 IN IP4 80.232.37.178
>   s=-
>   t=0 0
>   m=audio 47676 RTP/AVP 8 0 18 101
>   c=IN IP4 213.50.91.3
>   a=fmtp:18 annexb=yes
>   a=rtpmap:101 telephone-event/8000
>   a=fmtp:101 0-15
>   a=sqn: 0
>   a=cdsc: 1 audio RTP/AVP  8
>   a=cdsc: 2 image udptl t38
>   a=cpar: a=T38FaxUdpEC:t38UDPRedundancy
>   a=cpar: a=T38FaxVersion:0
>   a=cpar: a=T38MaxBitRate:14400
>   a=sendrecv
> 
> Regards
> Mark
> 
> On Mon, May 31, 2010 at 4:29 PM, David Ponzone <david.ponzone at gmail.com> wrote:
>> Mark,
>> This looks like a T38 Re-INVITE, but a weird one.
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>> 
>> 
>> 
>> Le 31/05/2010 ? 16:12, Mark Campbell-Smith a ?crit :
>> 
>> Hi All,
>> 
>> I'm sure I've discussed this before, but I searched through my gmail
>> and google and couldn't find the answer.
>> 
>> Below is the SDP parameters from my sip provider.  Is it mandatory to
>> always include the rtpmap details for PCMU/PCMA codes?  For example
>> something like 'a=rtpmap:8 PCMA/8000' and 'a=rtpmap:0 PCMU/8000'?
>> 
>> I'm using sipdroid on android and it rejects this with 'codec not supported'
>> 
>> Thanks
>> 
>> 
>> 
>> o=Sippy 141730476 0 IN IP4 xxx.xxx.xxx.xxx
>> s=-
>> t=0 0
>> m=audio 47676 RTP/AVP 8 0 18 101
>> c=IN IP4 xxx.xxx.xxx.xxx
>> a=fmtp:18 annexb=yes
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>> a=sqn: 0
>> a=cdsc: 1 audio RTP/AVP  8
>> a=cdsc: 2 image udptl t38
>> a=cpar: a=T38FaxUdpEC:t38UDPRedundancy
>> a=cpar: a=T38FaxVersion:0
>> a=cpar: a=T38MaxBitRate:14400
>> ]



From peter.olsson at visionutveckling.se  Mon May 31 09:41:20 2010
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 31 May 2010 18:41:20 +0200
Subject: [Freeswitch-users] RE FER with Replaces
In-Reply-To: <AANLkTil01_notTAD9lk7VBNJ_vMClYYgNh2uKL0V4rWV@mail.gmail.com>
References: <28712663.post@talk.nabble.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C567C9C8B2C@cooper>,
	<AANLkTil01_notTAD9lk7VBNJ_vMClYYgNh2uKL0V4rWV@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C567CA606F5@cooper>

uuid_simplify sends REFER with replaces-tags, if it's successful, the call will be completly removed from FS. So I guess in this case it's not accepted by the other ends of the legs. Check the log and see what happens.

You can also try deflect app, it will accept something like sip:xxxx at a.b.c.d. The difference is you only have one leg, and then send deflect to that leg, which will connect it to that destination.

I have tried REFER with about 15 different PBX'es, and I think about 5-6 of them handles it correctly, so it's not handled that well by all vendors.. :(

/Peter

________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Black Star [ewin.hogan at gmail.com]
Skickat: den 31 maj 2010 17:54
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] RE FER with Replaces

sorry if I missunderstood some concepts on a call but I originate two legs one of that legs isn't registered in FS, so I have two channels, then I bridge the legs and then I have the same two channels and a call, so then I use uuid_simplify and the call it's gone but I still have the two channels... it's that right? I looking for a kind of deflect that I have one leg and I deflect to for exaple to "sofia/internal/example at 192.168.1.XXX"  and then the call it's gone and also the channels I used too. well hope some one can help me :)

2010/5/31 Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>>
Use uuid_simplify on the bridged call, it will do exactly this.

/Peter


-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] F?r blacked
Skickat: den 29 maj 2010 00:45
Till: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
?mne: [Freeswitch-users] RE FER with Replaces


hi everyone well about my last post now I'll be more specific did anyone know
how can I make a transfer, a REFER with replaces to repleace FS in the loop
of the call and let the two legs with out FS.

hope someone can help me
--
View this message in context: http://old.nabble.com/REFER-with-Replaces-tp28712663p28712663.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.


_______________________________________________
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From infos at madovsky.org  Mon May 31 09:50:39 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 31 May 2010 12:50:39 -0400
Subject: [Freeswitch-users] alias folder
Message-ID: <9E374BCD682849C88F6008D62A4BD094@MOBILEE1705>

Hi,

is it a problem to have an alias folder (ex default directory in conf/directory) ?
because it seems that reloadxml doesn't work or maybe it's a permission problem...

Thanks

Franck
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From infos at madovsky.org  Mon May 31 10:06:57 2010
From: infos at madovsky.org (Madovsky)
Date: Mon, 31 May 2010 13:06:57 -0400
Subject: [Freeswitch-users] alias folder
Message-ID: <E85472D6772640648087C38DC8BC4074@MOBILEE1705>

found the problem
apparently FS didn't restarted well. I removed the pid and started again 
and everything is fine.

Thanks
  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, May 31, 2010 12:50 PM
  Subject: alias folder


  Hi,

  is it a problem to have an alias folder (ex default directory in conf/directory) ?
  because it seems that reloadxml doesn't work or maybe it's a permission problem...

  Thanks

  Franck
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From steveu at coppice.org  Mon May 31 10:42:28 2010
From: steveu at coppice.org (Steve Underwood)
Date: Tue, 01 Jun 2010 01:42:28 +0800
Subject: [Freeswitch-users] rtpmap question PCMU/PCMA
In-Reply-To: <088D4AC5-BE1D-4304-8139-FB00CFC796F9@jerris.com>
References: <AANLkTiloOAaoCWFD74aZQYuRex-kRCTG0zWLkV927jqu@mail.gmail.com>	<0495C809-A9F9-425D-ACF1-1B706324A1AF@gmail.com>	<AANLkTil6MLm29u_tB-BXL_V-geDrn0_1CUmlmDHGZ1N2@mail.gmail.com>
	<088D4AC5-BE1D-4304-8139-FB00CFC796F9@jerris.com>
Message-ID: <4C03F504.5070209@coppice.org>

Hi,

I think its a perfectly reasonable invite, including the shiny new 
capabilities stuff which should reach full RFC status shortly. As it is 
new, I think the jury is currently out on whether existing poorly 
implemented SIP packages will choke on it.

Steve


On 06/01/2010 12:35 AM, Michael Jerris wrote:
> This seems to be a badly broken sdp attempting to offer audio and t.38 but missing the m=image line from the sdp.
>
> Mike
>
> On May 31, 2010, at 10:57 AM, Mark Campbell-Smith wrote:
>
>    
>> Hi David,
>>
>> Its an INVITE.  Full invite below:
>>
>>    INVITE sip:gw+Phonzo at 124.xxx.xxx.xx:5080;transport=udp;gw=Phonzo SIP/2.0
>>    Record-Route:<sip:80.232.37.178;ftag=262787ae2a9104a0c7700794a69028aco;lr>
>>    Via: SIP/2.0/UDP
>> 80.232.37.178;branch=z9hG4bK8a48.ecc54191b911cf6bfd73daeefeae0ada.0
>>    Via: SIP/2.0/UDP
>> 80.232.37.178:5061;branch=z9hG4bK443b032c2a924807acf39718946e2c9e;rport=5061
>>    Max-Forwards: 16
>>    From: 010711xxxx
>> <sip:711xxxx at 80.232.37.178>;tag=262787ae2a9104a0c7700794a69028aco
>>    To:<sip:4610xxxxxxx at 80.232.37.178>
>>    Call-ID: M2Q1NjhhNmZjOGJjMDc3ODhlNzUyYzRiM2ZkMjQyZTE.
>>    CSeq: 200 INVITE
>>    Contact: Anonymous<sip:80.232.37.178:5061>
>>    Expires: 300
>>    User-Agent: Sippy
>>    cisco-GUID: 1658214937-1822691807-2631794736-96895450
>>    h323-conf-id: 1658214937-1822691807-2631794736-96895450
>>    Content-disposition: session
>>    Content-Length: 364
>>    Content-Type: application/sdp
>>
>>    v=0
>>    o=Sippy 141730476 0 IN IP4 80.232.37.178
>>    s=-
>>    t=0 0
>>    m=audio 47676 RTP/AVP 8 0 18 101
>>    c=IN IP4 213.50.91.3
>>    a=fmtp:18 annexb=yes
>>    a=rtpmap:101 telephone-event/8000
>>    a=fmtp:101 0-15
>>    a=sqn: 0
>>    a=cdsc: 1 audio RTP/AVP  8
>>    a=cdsc: 2 image udptl t38
>>    a=cpar: a=T38FaxUdpEC:t38UDPRedundancy
>>    a=cpar: a=T38FaxVersion:0
>>    a=cpar: a=T38MaxBitRate:14400
>>    a=sendrecv
>>
>> Regards
>> Mark
>>
>> On Mon, May 31, 2010 at 4:29 PM, David Ponzone<david.ponzone at gmail.com>  wrote:
>>      
>>> Mark,
>>> This looks like a T38 Re-INVITE, but a weird one.
>>> David Ponzone  Direction Technique
>>> email: david.ponzone at ipeva.fr
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>> Service Client IPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr  -   www.ipeva-studio.com
>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>
>>>
>>>
>>> Le 31/05/2010 ? 16:12, Mark Campbell-Smith a ?crit :
>>>
>>> Hi All,
>>>
>>> I'm sure I've discussed this before, but I searched through my gmail
>>> and google and couldn't find the answer.
>>>
>>> Below is the SDP parameters from my sip provider.  Is it mandatory to
>>> always include the rtpmap details for PCMU/PCMA codes?  For example
>>> something like 'a=rtpmap:8 PCMA/8000' and 'a=rtpmap:0 PCMU/8000'?
>>>
>>> I'm using sipdroid on android and it rejects this with 'codec not supported'
>>>
>>> Thanks
>>>
>>>
>>>
>>> o=Sippy 141730476 0 IN IP4 xxx.xxx.xxx.xxx
>>> s=-
>>> t=0 0
>>> m=audio 47676 RTP/AVP 8 0 18 101
>>> c=IN IP4 xxx.xxx.xxx.xxx
>>> a=fmtp:18 annexb=yes
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-15
>>> a=sqn: 0
>>> a=cdsc: 1 audio RTP/AVP  8
>>> a=cdsc: 2 image udptl t38
>>> a=cpar: a=T38FaxUdpEC:t38UDPRedundancy
>>> a=cpar: a=T38FaxVersion:0
>>> a=cpar: a=T38MaxBitRate:14400
>>> ]
>>>        
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>    



From ewin.hogan at gmail.com  Mon May 31 11:29:33 2010
From: ewin.hogan at gmail.com (Black Star)
Date: Mon, 31 May 2010 13:29:33 -0500
Subject: [Freeswitch-users] RE FER with Replaces
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C567CA606F5@cooper>
References: <28712663.post@talk.nabble.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C567C9C8B2C@cooper>
	<AANLkTil01_notTAD9lk7VBNJ_vMClYYgNh2uKL0V4rWV@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C567CA606F5@cooper>
Message-ID: <AANLkTimeiQ4rXGvYOnh_tPXEvx4UW0HEl_u1LX0KnaIQ@mail.gmail.com>

so it's not working the refer if I still having two channels active right?

well thank you so much for your time I noticed somethings I didn't :)

Greetings from Mexico
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From mcampbellsmith at gmail.com  Mon May 31 12:15:35 2010
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Mon, 31 May 2010 21:15:35 +0200
Subject: [Freeswitch-users] rtpmap question PCMU/PCMA
In-Reply-To: <4C03F504.5070209@coppice.org>
References: <AANLkTiloOAaoCWFD74aZQYuRex-kRCTG0zWLkV927jqu@mail.gmail.com>
	<0495C809-A9F9-425D-ACF1-1B706324A1AF@gmail.com>
	<AANLkTil6MLm29u_tB-BXL_V-geDrn0_1CUmlmDHGZ1N2@mail.gmail.com>
	<088D4AC5-BE1D-4304-8139-FB00CFC796F9@jerris.com>
	<4C03F504.5070209@coppice.org>
Message-ID: <AANLkTilnkdVkhRRUbfezjyThErvtSpkHL8eASnoU4u5-@mail.gmail.com>

Thanks Guys.

I should have noted that this is not related to an FS fault at all.
When using FS in bypass media mode, the call is rejected by the
client.  Without bypass media mode it works.

I just know there is a huge sip knowledge on this mailing list, and
would be able to get my answers easily.

I guess the android client sipdroid does not like the broken and/or
shiny new SDP format.

Cheers

On Mon, May 31, 2010 at 7:42 PM, Steve Underwood <steveu at coppice.org> wrote:
> Hi,
>
> I think its a perfectly reasonable invite, including the shiny new
> capabilities stuff which should reach full RFC status shortly. As it is
> new, I think the jury is currently out on whether existing poorly
> implemented SIP packages will choke on it.
>
> Steve
>
>
> On 06/01/2010 12:35 AM, Michael Jerris wrote:
>> This seems to be a badly broken sdp attempting to offer audio and t.38 but missing the m=image line from the sdp.
>>
>> Mike
>>
>> On May 31, 2010, at 10:57 AM, Mark Campbell-Smith wrote:
>>
>>
>>> Hi David,
>>>
>>> Its an INVITE. ?Full invite below:
>>>
>>> ? ?INVITE sip:gw+Phonzo at 124.xxx.xxx.xx:5080;transport=udp;gw=Phonzo SIP/2.0
>>> ? ?Record-Route:<sip:80.232.37.178;ftag=262787ae2a9104a0c7700794a69028aco;lr>
>>> ? ?Via: SIP/2.0/UDP
>>> 80.232.37.178;branch=z9hG4bK8a48.ecc54191b911cf6bfd73daeefeae0ada.0
>>> ? ?Via: SIP/2.0/UDP
>>> 80.232.37.178:5061;branch=z9hG4bK443b032c2a924807acf39718946e2c9e;rport=5061
>>> ? ?Max-Forwards: 16
>>> ? ?From: 010711xxxx
>>> <sip:711xxxx at 80.232.37.178>;tag=262787ae2a9104a0c7700794a69028aco
>>> ? ?To:<sip:4610xxxxxxx at 80.232.37.178>
>>> ? ?Call-ID: M2Q1NjhhNmZjOGJjMDc3ODhlNzUyYzRiM2ZkMjQyZTE.
>>> ? ?CSeq: 200 INVITE
>>> ? ?Contact: Anonymous<sip:80.232.37.178:5061>
>>> ? ?Expires: 300
>>> ? ?User-Agent: Sippy
>>> ? ?cisco-GUID: 1658214937-1822691807-2631794736-96895450
>>> ? ?h323-conf-id: 1658214937-1822691807-2631794736-96895450
>>> ? ?Content-disposition: session
>>> ? ?Content-Length: 364
>>> ? ?Content-Type: application/sdp
>>>
>>> ? ?v=0
>>> ? ?o=Sippy 141730476 0 IN IP4 80.232.37.178
>>> ? ?s=-
>>> ? ?t=0 0
>>> ? ?m=audio 47676 RTP/AVP 8 0 18 101
>>> ? ?c=IN IP4 213.50.91.3
>>> ? ?a=fmtp:18 annexb=yes
>>> ? ?a=rtpmap:101 telephone-event/8000
>>> ? ?a=fmtp:101 0-15
>>> ? ?a=sqn: 0
>>> ? ?a=cdsc: 1 audio RTP/AVP ?8
>>> ? ?a=cdsc: 2 image udptl t38
>>> ? ?a=cpar: a=T38FaxUdpEC:t38UDPRedundancy
>>> ? ?a=cpar: a=T38FaxVersion:0
>>> ? ?a=cpar: a=T38MaxBitRate:14400
>>> ? ?a=sendrecv
>>>
>>> Regards
>>> Mark
>>>
>>> On Mon, May 31, 2010 at 4:29 PM, David Ponzone<david.ponzone at gmail.com> ?wrote:
>>>
>>>> Mark,
>>>> This looks like a T38 Re-INVITE, but a weird one.
>>>> David Ponzone ?Direction Technique
>>>> email: david.ponzone at ipeva.fr
>>>> tel: ? ? ?01 74 03 18 97
>>>> gsm: ? 06 66 98 76 34
>>>> Service Client IPeva
>>>> tel: ? ? ?0811 46 26 26
>>>> www.ipeva.fr ?- ? www.ipeva-studio.com
>>>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>>>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>>>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>>>> d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il
>>>> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
>>>> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>>>>
>>>>
>>>>
>>>> Le 31/05/2010 ? 16:12, Mark Campbell-Smith a ?crit :
>>>>
>>>> Hi All,
>>>>
>>>> I'm sure I've discussed this before, but I searched through my gmail
>>>> and google and couldn't find the answer.
>>>>
>>>> Below is the SDP parameters from my sip provider. ?Is it mandatory to
>>>> always include the rtpmap details for PCMU/PCMA codes? ?For example
>>>> something like 'a=rtpmap:8 PCMA/8000' and 'a=rtpmap:0 PCMU/8000'?
>>>>
>>>> I'm using sipdroid on android and it rejects this with 'codec not supported'
>>>>
>>>> Thanks
>>>>
>>>>
>>>>
>>>> o=Sippy 141730476 0 IN IP4 xxx.xxx.xxx.xxx
>>>> s=-
>>>> t=0 0
>>>> m=audio 47676 RTP/AVP 8 0 18 101
>>>> c=IN IP4 xxx.xxx.xxx.xxx
>>>> a=fmtp:18 annexb=yes
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-15
>>>> a=sqn: 0
>>>> a=cdsc: 1 audio RTP/AVP ?8
>>>> a=cdsc: 2 image udptl t38
>>>> a=cpar: a=T38FaxUdpEC:t38UDPRedundancy
>>>> a=cpar: a=T38FaxVersion:0
>>>> a=cpar: a=T38MaxBitRate:14400
>>>> ]
>>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> http://www.freeswitch.org
>