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